similar to: Page minimum number of extensions

Hi, I'd like to implement some way of controlling remote SIP clients while in a call, to execute remote commands. The call topology (think of a PA system) is this: * the caller is in a MeetMe() conference room * the callees are Page()d, then the dynamic conference room is connected to the previous one I'm wondering if Asterisk is relaying DTMF (SIP info or RTP) from the caller to the

Hi, I'm struggling with a feature in my home phone setup. I have several phones using both SIP and SCCP. What I try to do is to create a dynamic feature that works similar to the blindxfer feature built into Asterisk. What I want is the possibility for the called part to push a number sequence (for example *#) to redirect the callee to a fixed extension or (for example *123#) to redirect the

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

I have a "support call" AGI script that has been working flawlessly for a couple of years now. It dumps the customer into a MeetMe conference room, then dials a bunch of support engineers, and connects anyone who accepts the call into the conference room. The conference room is recorded. After the support call is over, the recording is emailed to a list for quality control, etc. It

Hi, We are using Vicidial and sometime even when agent disconnects, outgoing call originated by dialer is still active. Since call was initiated by dialer and then bought into meetme conference of agent and we can't corelate this call to any agent channel. When agents are dialing, channels doesn't show calls vicidial2*CLI> show channels Channel Location

Hi asterisk-users, We are using Asterisk 1.2.12.1, and are trying to use the Page application. It seems to work but after approx 4-5 seconds the call is hung up. The dialplan code look like this: exten => _*2XX,1,AGI(get-paging-devices.agi,${EXTEN:2}) exten => _*2XX,n,GotoIf($[ "${PAGING_DEVICES}" = "invalid" ]?i,1) exten => _*2XX,n,SIPAddHeader(Call-Info:

I've just noticed a strange behaviour with a MeetMe conference. I have a pair of phones (GS BT102) on my desk, and dialled both of them into a conference on speakerphone. If I spoke or made a sound, I heard it replayed from both speakers together a split second later, as expected. I went away for about 15 minutes, leaving the conference running. When I came back any sound I made came back

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

/Hello Users, Good Morning, In Conferemcing How to Disconnect the phone while in between the Conference ..... When *I press the ' # ' key for Disconnecting the Conference.......... Below the Following to shows some Warning, ( in Red Color ) from-sip en *CLI> -- Executing Playback("SIP/9002-08f9feb8", "conf-hasentered") in new stack

I'm trying to use a simple page function. It starts a MeetMe conference with the devices I've listed, but the devices hang up after 3-5 seconds. After doing some research I found this was a problem, and I needed to remove a (5) from app_page.c Well, my app_page.c didn't have the (5). I did make clean; make install again just in case I had some weird compiled version installed that

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"

Hi all, there is an application called changrab with quite interesting capabilities: http://www.freeswitch.org/asterisk_stuff/app_changrab.c I think it is available in the 1.4 version by default!? This application can connect to channels which are already UP. The only possibility AFAIK to connect channels after they are UP are the well known conferencing applications meetme and conference. If

I previously had Asterisk 1.0.7 running on a Linux 2.4.x kernel with ztdummy. I was able to do things like meetme and music on hold. I recently installed Asterisk 1.0.9 on a different machine with a Linux 2.6.x kernel running ztdummy. I installed and configured everything the same way, but when I try to call into a conference room I get the error message stating, "that is not a valid

On late august, there was a thread about setting up some meetme conferences to be able to follow Astricon remotely. This indeed could be nice for those that can't attend for various reason. And of course is a demonstration of Asterisk capabilities... :) (Astricon without a remote conference for guest is like a big it expo without internet connections...) I have some bandwidth here, so can

I have successfully set up a conference room on my asterisk server, I have been trying to make the 'M' for music on hold option work (when the first person enters the room they are told they are the first and then they are supposed to hear music on hold) but it didn't matter which way I wrote it this feature wouldn't work. Basically it wouldn't allow the conference to be

Hi - I found the "maxusers" defined in meetme.c, but I'm not sure how this value is set. Does anybody know if one can limit the number of users permitted in a meetme conference? I know there's MeetmeCount(), but I'd rather avoid the dialplan logic and just set maxusers instead. Thanks, Noah

Greetings, I have created simple conferencing solution before using meetme application, but this times its a little tricky. My client needs a functionality to call multiple extensions to join a conference room. Extensions will ring like in a ring group, and on pick up, user will be either automatically added to the conference room, or maybe I'll program them to enter 9 to accept and 8 to

Hello, I'm working on some dialplan rules to pull multiple users into a conference call. I have some fairly straightforward rules which start up a new MeetMe conference, allow escape with the * key to invite more users, then use a features.conf sequence to bring the new user into the conference with ChannelRedirect. The problem I'm running into is the time in the MeetMe conference

One of our users recently had a powerfail while connected to our meetme gateway. (Asterisk 1.4.17 on debian 4.0) Through the course of it, asterisk never hung up. His system came back up, and started sending ICMP port unreachables, but the stream went on, flooding him with "silence" media stream packets (there was nobody else in the conference). Is asterisk aware of ICMP

hi, i write this looking for free conference room, i checl code and don?t see any error but die at priority 7 if room 1001 have users in exten => _1NXXNXXXXXX,1,RouteCall(${EXTEN}) exten => _1NXXNXXXXXX,2,GotoIf($[${DESTINATION1:0:3} = CONF]?3:13) exten => _1NXXNXXXXXX,3,Setvar,var=0 exten => _1NXXNXXXXXX,4,MeetMeCount(1001|var) exten => _1NXXNXXXXXX,5,GotoIf($[${var} =0]?7:6)