similar to: Location of MeetMe Recordings

Hello, Is there a way to change the format from the default? 'r' - Record conference (records as ${MEETME_RECORDINGFILE} using format ${MEETME_RECORDINGFORMAT}). Default filename is meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav. - requires chan_zap.so Many thanks ******************************************************************** This email and any attachments

Hi i am experiencing this problem: MEETME_RECORDINGFILE=/data/asterisk_data/_${DATETIME}_CONFERENCE exten => 9999,1,MeetMe(666|1Arxq) exten => 9998,1,MeetMe(666|1Axq) exten => 9997,1,MeetMe(666|1xq) I make a conference between 3 person dialing A dials 9999 B dials 9998 C dials 9997 all works fine but the datetime won't be updated, it still remain for example 13:40 until i do a

I am having a weird issue with setting the recording file for the Page app. Here is some quick background info I have a macro that pages all my phones: [macro-pageall] ; Context for paging all devices. ; This will search the sip table in the realtime database ; for all phones that start with a number. That number is ; passed to this macro as ${ARG1}. ; ; ARG1 = The

Hi all, I have an external application commanding asterisk by AMI and AsyncAGI. I also have a dialplan like this: ; AsyncAGI extensions exten => _8.,1,Noop(entering in AGI loop at 8 ${EXTEN}); exten => _8.,n,AGI(agi:async); exten => _8.,n,Hangup(); ; Meetme extensions exten => _1.,1,Noop(Conference ${EXTEN} ${CONTEXT}); exten =>

Hi there, I'm using asterisk meetme function like: exten => 9070,n,MeetMe(|dcM) and everything works pretty well. But I would like to add a review of the entered conference number before the user jumps into the conference. Somthing like: *:"Please enter the conference number followed by the hash key" (works) U: 123456# (works) *: "You are entering conference number

I am looking to see if anyone has seen this problem before. I am setting the MEETME_RECORDINGFILE variable in a macro, then using the r option with the Page application to record the page. But the page is only recorded to the file specified in MEETME_RECORDINGFILE sometimes... Sometimes it works and sometimes it doesn't. When it doesn't work it places the recorded file in

Hi iam using 1.2.17 does any one have information meetme in realtime and store in mysql i dont see any document could some one help me is this possible ? ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070606/36d236c2/attachment.htm

Hello , I am new to Asterisk. Trying to use Meetme for Audio Conferencing. Got Zaptel card etc. and i could see app_meetme.so nicely loaded. Now : 1. how to start a conference ? 2. how to add a user ? 3. How can a user join a conference ? After looking at certain links on Net I tried to

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

Any thoughts on the following? I am running asterisk from CVS (downloaded yesterday's version, just to be sure) on a test system with no digium cards in it, so I have installed ztdummy (see logs and screenshots below) as a timing source. When I call the music on hold extension from a Sipura Sip connected analog phone, I hear nothing and start getting Warning[98310]: chan_sip.c:674

Stay away from Alliance Systems. We ordered $15k worth of Polycom's over a month ago and we're still waiting. Our account rep's communication with us on what the delay has been, has been terrible. Doug. -----Original Message----- From: Gavin Adams [mailto:me@gavinadams.org] Sent: Friday, January 27, 2006 8:26 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion'

running Asterisk 1.8.9.0-rc2, what are the ways to interface with ConfBridge ? I see the CLI command 'confbridge' documented for asterisk 10, but i dont see how to interface with confbridge on 1.8 What I'm trying to do is keep track of conferences that are used. I tried something like the below, but not only does Confbridge not return, but i'd need something that erases the

Hi, i am using the n-way-call dialplan solution found on voip-info. i have added its entry in applicationmap of features.conf file. the problem is......its not working. to activate the n-way call i dial *0 but nothing happens. i have played around with dtmf and codec settings but no success. the extensions and sip configuration is below if you want to have a look. I dont have any clue why its not

Hi there, I know this question is kind of stupid. But, I don't know anywhere else to ask. I've received some answers when I asked about the need of having a zaptel interface to make the meetme application work, that said that it was better to have a real hardware then the zaptelrtc software modules. So, my question is, would any of the following cards work with the meetme

Hi, I am trying to achieve 3-way conferencing taking hint from wiki link http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO Here is the scenario: 1. user "ua1" calls user "ca1" 2. "ua1" then presses the feature code "*0" to redirect "ca1" to conference room 300 3. "ua1" then dials the user "33" 4. user

Hi How to set a threeway conference in asterisk only for VOIP (I am using only SIP channel). Thanks Nikhil

Hi all, I'm trying to get the MeetMe system to take a caller and announce to them they've joined the conference in addition to the other members of the conference assuming previous members of the conference >= 1. I can see where the meetme.c app actually processes it using the ast_pthread_create_background(&conf->announcethread, NULL, announce_thread, conf); function. The

Hi, I didn't notice on my first answer, but we are on the -dev list and this is not related to asterisk code developing. I will answer you on the -users list, so we can continue the discussion there. Cheers, -- Ing. Miguel Molina Grupo de Tecnolog?a Millenium Phone Center Anahi Ludue?a escribi?: > Hi, thanks Miguel. > I have another question: if I want to call the GoSub

Hi, I am not sure if I am doing something wrong, but I can't get MeetMe to work with GSM codec (Asterisk 1.6.1 SVN r190371). My config files below: ---- sip.conf: ---- [general] context=common canreinvite=no bindport=5060 bindaddr=78.105.1.127 disallow=all allow=alaw allow=gsm rtptimeout=600 rtpholdtimeout=3600 rtpkeepalive=30 nat=no jbenable=yes tcpenable=no realm=dev-sip.wima.co.uk

I'm having problems getting an Xorcom USB Bri 4 dialling out in Australia. I can receive calls into the system without an issue, but I can not for the life of me dial out of the system. Below are my configs, I'm hoping its something simple that I just can't see as I've been looking at it for to long. Can any one point me in the right direction. P.S. Yes it is meant to be in TE