similar to: Invite somebody to a conf call

Hello I'm looking for a wayy to modify extensions.conf It seems that PutConfig AMI command is not supposed to work on extensionsq.conf Any ideas? Thanks Vadim

Date: Mon, 23 Jun 2008 08:00:08 -0400 From: "David Backeberg" <dbackeberg at gmail.com> Subject: Re: [asterisk-users] Replace music-on-hold on MeetMe with ringing sound To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Message-ID: <3de056a30806230500k7e66185l7bfe473ed398ebf6 at

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

Hi guys, I need to realize a sort of automatic call monitoring dialplan. This is exactly what I need: - A user originate a call - When the call is bridged (or just before) I need to invite automatically a third party to the conversation that should hear the audio channel but not speak (it's a monitoring application for a callcenter) The person in charge of monitoring cannot use ChanSpy or

Hi guys, I need to realize a sort of automatic call monitoring dialplan. This is exactly what I need: - A user originate a call - When the call is bridged (or just before) I need to invite automatically a third party to the conversation that should hear the audio channel but not speak (it's a monitoring application for a callcenter) The person in charge of monitoring cannot use

Hi, Is there anyway to have a MeetMe conference start only when somebody (anyone, let's say I don't want to manage who is the "marked user") connects and has the admin PIN instead of the user PIN? I would have assumed this was an obvious feature, but I dont see it on the Wiki. Or I am misreading it, which is entirely possible. Mike -------------- next part --------------

Hello. I have a strange problem with the MeetMe application. Configured is a misdn msn to go into a preconfigured MeetMe room. exten => 12,1,MeetMe(1234,pIM) The first caller gets the prompt to enter the pin and then gets connected to the MeetMe room. The second caller gets also the prompt but after entering the right key he hears a dialtone followed by the message: The number you have

On Thursday 21 July 2005 17:42, asterisk-users-request@lists.digium.com wrote: > > > Thanks Adam. This helps some, but I'm still not sure how you mean ? > > > for me to acheive 1). I would have to perform a Dial-command no matter > > > what, so I guess I would have to make an interruption from the > > > manager API, but I don't manage to find a command

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)

With 1.6.1, I'm trying to set up a test of meetme for creating dynamic conferences. cat meetme.conf [rooms] conf => 600 extensions.conf: [meetme] exten => 2663,1,MeetMe(,D) exten => 2663,n,Hangup() exten => 2666,1,MeetMe() exten => 2666,n,Hangup() What I'm expecting is to dial 2663, get a conference room number ( 600, I suppose since it's the only room ), and set

My system is linked to a legacy PBX via Q-SIG and most of my tests so far have been from that PBX. I created a number of MeetMe conference rooms and they work fine when called from the legacy PBX. However, when there's a MeetMe room with a legacy caller and a SIP phone, the SIP phone can hear the legacy caller. But the legacy caller can't hear the SIP phone. However, "meetme show

Hello, To make my dialplan more robust, I thought I wouldn't include any meetme-specific rules and I would exlusively rely on meetme.conf data. For each dialed number, I would check if this number is used as a conference room number in meetme.conf. When I'm trying to implement this, I can see that : 1. AST_CONFIG is not convenient to parse lines like "conf=>1234", as

Hi, today i have taken a strong look at meetme.c what i am trying to accomplish is the following: it should be possible to access an menu from within the conference in order to perform special tasks, eg. to dial another number so that the called person is joined with the conderence. my first try was to use an agi-script for this, but as with agi enabled sip-channels (for which

Hi, the following szenario leads to a problem: I connect an CAPI channel to an AGI-Script per Manager API. This Agi script starts the MeetMe-Application. The Person on the Capi Channel is now able to speak with the other conferess in the MeetMe-Room. But if the CAPI channel hangs up, the busy tone is streamed into the MeetMe Room for several seconds, until the CAPI HANGUP-Signal is finally

I can't seem to locate any documentation on what this does. I tested it out with a simple static conference room: exten => conference,1,MeetMe(,1aMqw) and a static room defined in meetme.conf: conf => 123456,22,1 Users can get in with either of the pins, but I don't see that it does anything - I can't access the admin menu, nor does it set the user as marked to open up the

Hello there! Somebody tried the meetme|b which initiates the conf-background AGI. Actually I want to originate another call from a conference.my AGI originates the call and connects it to the conference, but the calleeee is nowhere My extension exten => 21,1,meetme(21|pb) and my AGI **************************************************************************** #!/usr/bin/perl -w

Has anyone a solution for me - with "Meetme(,Ms)" asterisk plays "conf-invalid" if a room not exist - with "Meetme(123,Ms)" asterisk plays not "conf-invalid" if the room not exist and asterisk hangup I am happy about any proposal. Thanks Daniel

Hi, I'm new to asterisk and have a little trouble in developing my first more complex dialplan. The basic task is a click to call solution: - call one number via sip, play some announcements, do cdr etc. and put the callee into an conference room with music on hold - call a second number via sip, play some announcements, do cdr etc. put the callee into the same conference - have a nice chat

Hi Peoples I am having problems with meetme, in that it responds with "conf-invalid" when I dial a conference number. I notice that there is a note with regards to ztdummy, and the need for that to be loaded. Is this still the case? Is meetme dependent on this module? I do NOT use zaptel cards in my system, and there for zaptel is not loading. Can anyone shed some light

Hi All, I use the Asterisk Manager Interface to redirect the channels. There have two channels : SIP/voip_out_22-809c (None) Up Bridged Call(SIP/612-5456) SIP/612-5456 s@macro-monitor:10 Up Dial(SIP/0882@voip_out Then I send a redirect request like below : Action: Redirect Channel: SIP/612-5456 ExtraChannel: