similar to: Meetme Options (new one)

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:

Hello, all. I must be brain cramping badly on our Asterisk 1.6.1.1 installation. Our MeetMe macros are working fine except they do not prompt for a PIN. So I made a very simple conference room: exten => 7777,1,MeetMe(123456,cMaAsx,123456) Shouldn't this prompt the user who dials 7777 to enter a PIN before entering the conference room whether or not a PIN is defined in meetme.conf? I

Hello, all. I've been playing with MeetMe and talker optimization seemed like a great idea. I activated it as follows: exten => 201,1,MeetMe(100201,cTo) However, although I can see who is the talker on the CLI pbx01*CLI> meetme list 100201 User #: 01 1001 Denise Dion-Sullivan Channel: SIP/1001-1e1db7c8 (not talking) 00:00:33 User #: 02 1000 John A. Sullivan III

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:

Is there a UK Asterisk users group? Would be interested in contacting others in the UK who use asterisk for either home or business applications. If there is, could someone provide me with some contact details, else anyone who's also interested, contact me off list. Cheers, Ben Merrills -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is there a way of getting the dialled number from an AGI? Is it passed in the initial variables, or can it be pulled out or passed across from the dial plan? Cheers, Ben Merrills Griffin Internet -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040630/71e0bcda/attachment.htm

I was wondering if anyone knew how to do the following.... Call comes in, gets put into a Queue, say `Sales`. Then the queue member is presented with the option to exit the queue, yet have the phone system sit in their place for them. When the virtual member reaches the front, call back the caller and connect them to the agent. Any ideas? Did i explain that ok? :) Cheers, Ben Merrills

Is there a way to log all PRI events to a logfile? Cheers, Ben Merrills Griffin Internet T: 0870 8040862 F: 0870 8040805 W: www.griffin.com <http://www.griffin.com/> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041123/d983141f/attachment.htm

Has anyone tried integrating Asterisk and Rhetorical's rVoice software? We're evaluating different approaches to system announcements via T2S. Has anyone gone down this route that could give some advice? I've installed festival and wasn't too impressed, the demo one the website seems far better quality and clarity then the defaults in the source package. However I must admit

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

hi guys: recently I want to buinding a meeting confence on asterisk and use the meetme application. I have a ztdummy kernel afteri the lsmod commond: ztdummy 5768 0 zaptel 182660 28 zttranscode,ztdummy crc_ccitt 3008 1 zaptel I also configure the meetme.conf conf => 1000; my extensions.conf [default] exten =>

-----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Tuesday, May 25, 2004 5:30 AM To: asterisk-users@lists.digium.com Subject: Asterisk-Users digest, Vol 1 #3891 - 8 msgs Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To subscribe or

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 =>

One of the features that I thought would be popular with the Web-MeetMe suite is the ability to start all non-admin callers in a muted state and selectively unmute them. For example any large conference that is of an announcment nature with a Q&A session. It's probably a feature I should have tested better, but I just discovered that a caller that is joined to a MeetMe with the |m flag

Hi, If there are 5 asterisk servers on the local net and each server runs meetme, eg. 3311,3321,3331,3341,3351 respectively. Can I connect these 5 meetme conferences to one meetme using IAX2? Regards, Zen

Hi, Is there and trick to getting a conference room up and running.. I have 'conf => 7500' in the meetme.conf file and 'exten => 7500,1,MeetMe(7500)' in the extensions.conf file (in the same context as my phone extensions).. When I dial extension 7500 I get the voice saying "That is not a valid conference number, Please try again.." <beep> <beep>

Hi, This is pretty much a repost but I still havent been able to get MeetMe to work.. I am using a Dev Kit lite.. so that should satisfy the Zaptel requirement for MeetMe.. meetme.conf looks like this.. [rooms] conf => 7500 In extensions.conf I have an [extensions] context and within that same context I have the line.. exten => 7500,1,MeetMe(7500) When I dial 7500 I get the message

Hello list, I wonder if someone might be able to clear up something for me. I recently set up asterisk and have now managed to get the MeetMe application up and running. When I dial the extension to access the conference/MeetMe application, the only prompt I hear is:"You are currently the only person in this conference." When I use a friend's newly installed asterisk, I hear:

I do not have any zaptel hardware on the Asterisk box, I could not have meetme functioning. I did modify the Makefile in zaptel directory on line 168 by including ztdummy as one of the modules to compile in. The error message from the concole: -- Executing MeetMe("SIP/1002-e9ca", "4700") in new stack == Parsing '/etc/asterisk/meetme.conf': Found

We'd be more than willing to test :) Our PRI provider has been trying to push us over to SS7 for ages. So we'd be very interested in assisting you. Ben -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Steve Underwood Sent: 12 January 2005 12:49 To: Asterisk Users Mailing List - Non-Commercial Discussion