similar to: question on asterisk 1.8 meetme

I am reading comments on the Wiki for meetme http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe from 2004 about how and AGI does work with non zap channels. Is this still valid 3 years later and 1.4.4? How do I bring people into a meetme and play a message to all of them when they are on SIP channels? Jerry

I am running asterisk 1.4.18 trying to use MeetMe and option b. I am getting permissions denied failed to execute conf-background.agi on the CLI lrwxrwxrwx 1 root root 37 Mar 17 10:11 conf-background.agi -> /home/silentm/bin/conf-background.agi my conf background is a symbolic link - then my permissions are : [root at devcentos5x64 src]# ls -l /home/silentm/bin/conf-background.agi

Hello fellow asterisk users, I've been facing a problem when using MeetMe's admin functionality to unmute users in a conference using *Asterisk 1.6.2.11*. I've tried: 1) MeetMeUnmute (AMI) 2) MeetMeAdmin(AMI) 3) MeetMeChannelAdmin(AMI) and also tried via console : "asterisk -rx 'meetme unmute conf_no user_no'" and the available AGI functions. but all of this to no

How do I stop the "you have been kicked by this conference" message from speaking? I first had MeetMe(conf, l) and I get the kicked message. I tried Meetme(CONF, lq) and I still get he kicked message. and it still says it. Thanks, Jerry

I'm taking inbound calls on an * server, then transferring them to a second * server where they join a MeetMe conference. If I have 'notransfer=yes' set on the first * server it works fine, but if I allow the transfer (call then shifts to be between the DID provider and the second server), the call is dropped 3-5 minutes later. There is no firewall on my end, and the two

I am looking for a way to automatically close a meetme conference when either a user hangs up or through an agi call? Some method that would automatically terminate the meetme. Is there a way to do that? Jerry

What does it take to get ztdummy to work correctly? I have a new laptop HP HDX9200. I am running asterisk 1.4.19 and zaptel 1.4.9.2 Zaptel compiles fine. asterisk compiles fine. ztdummy loads asterisk runs. Problem is playback() does not work. So then I stop zaptel, asterisk runs and playback() now works. However, meetme()'s dont work. I need ztdummy I'm pretty sure for that. I am

I am hearing a you have been kicked from the conference message in asterisk 1.2.23. I dont want to hear that. I am using 1qt for the meetme. How can I disable that message? THanks Jerry

Hey Guys, We upgrade asterisk from 1.2.x to 1.8.2.3 and my one of agi script doesn't working We have allpage.agi script for paging system on all polycom 501 but after upgrade it broke. Any idea what is this error ? extension.conf exten => 7770,1,agi(allpage.agi) exten => 7770,2,meetme(7770,dq) exten => 7770,3,playback(beep) exten => 7770,4,hangup following is agi debug....

When I do "core show channels" on 1.8 it gives me something like: Channel Location State Application(Data) DAHDI/i1/3175551212- s at default:10 Up BackGround(SM_ATTENDANT) 1 active channel 1 active call 188 calls processed No active MeetMe conferences. What channel is i1?? It used to show me DAHDI/18/3175551212 . How do I

Steve, I am trying to get tx_fax to work. I am using a TDM2401E card. I have a 3 page fax and I only receive the first page on every attempt. I think I have enabled debug output below. Can you tell me what the problem might be? I am using snapshot from oct 26. asterisk 1.2.13 and libtiff 3.6.1-12 from redat/centos 4.4. THanks, Jerry --------- Oct 28 13:13:40 DEBUG[22763]: app_txfax.c:69

Hi All, Anyone out there successfully tested Asterisk 1.8 with Web-Meetme 4.0v in my case my asterisk got crashed when i dialing conf room number. Best, S -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20101215/52082964/attachment.htm>

Hi. After the latest upgrade of asterisk-1.8, I notice that meetme does not allow the menu, if musiconhold is active which only occurrs if a single user is in the conference. Sometimes I have to unmute someone because for various reasons they cannot do this and they are the first one in the conference. I looked at the code and its a one line change, and I wonder if it was done deliberately or

I am trying to use meetme() on SIP channels. I found this line on voip-info.org ------------- It *is* necessary either to have a Digium card or a dummy timing driver (e.g. ztdummy or zaprtc) in order for MeetMe to work at all, but that doesn't help you use AGI with SIP channels: They have no capacity to use any AGI script at all. If they try to, they get no audio. ----------------- I am

All, I have noticed that placing a call in the outgoing spool during a reload the call may fail. Try the call again after the reload is done and it will complete. This seems like a bug. During a reload calls should be suspended or something? Thoughts? Jerry

Jeremy wrote: > What kind of luck are people having with the Web-MeetMe control? The > condition of the page on the voip-info wiki makes me a bit nervous about > putting Web-MeetMe into a production environment. Use of MeetMe has > really taken off here since installation and I need a scheduling and > provisioning system for PIN numbers etc. Are there any other solutions > out

Hi. I am having a strange problem when using the 1.4 version of asterisk and zaptel. If I call from a pstn line into the asterisk box using a phone number which calls the box via sip, then once I am in the meetme conference nothing happens when I hit the star key -- I cannot get the user menu. There is nothing in the logs at all its as though asterisk never sees the digit at all. Now if I do

I am sending an email from my machine devcentos5x64. the transcript below (hangs for 60 seconds) at the line: MAIL From:<root at devcentos5x64.msgnet.com> SIZE=56 AUTH=root at devcentos5x64.msgnet.com The email succeeds - but I am trying to figure out the 60 second delay. Neither email server is busy. Nothing is waiting. the DNS on both machines point to the same nameserver. The DNS

To continue here with problems on this compaq v2000 laptop, I put kernel source on a USB disk and plugged it into the v2000. NOTHING IS recognized. I tried to manually mount the disk and nothing either... I thought USB was well established.... I thought trying to recompile the kernel for realtek support might get my networking going... I am stuck??? Jerry -------------- next part

Greetings List! I'm currently rolling out a new deployment of Asterisk 1.8 to replace existing 1.2 servers...and have run into an issue which could use your assistance! For testing I have trunked (iax2) two of the servers - one running 1.8 and the other at 1.2. Calls placed from SIP --> SIP sound fantastic and crystal clear. However, when I place a echo test call (*43) from 1.8 to 1.2