similar to: Asterisk transfer to a conference using feature code?

Emrah wrote: >> This is an asterisk-users question, and would have been more appropriate to have >> asked there. >> >> Instead of setting up your conferences in meetme.conf, you could set them up >> dynamically in the dialplan, and then you can control whether the user is >> prompted for a pin or not when joining the conference, based on whatever logic

Hi, I know this question may have been asked before (although the archives don't seem to suggest it), but has anyone had any problems with Asterisk accepting a PIN number for a conference room. At this point in time I have established the conference definition in the meetme.conf file as well as specifying the appropriate lines in the extensions.conf file. meetme.conf file: conf =>

Here in Sweden, it's supposed to be springtime. A wonderful time of the year, with sunny skies and wonderful weather. Almost summer. Today, it's not. It's winter all over again with rain and only 3 degrees celsius outside. Better to stay inside and write a weekly Asterisk newsletter :-) This week's topics: ------------------- * Looking beyond Asterisk 1.0/1.1 - what's up? *

All; What I want to do is create a way to easily send callers into a conference room to have an N-way conference call. I created an extension '100' that calls the MeetMe() command. Then all I need to do is transfer a caller using a blind transfer (*2 in my case) to extension 100. Then I can dial a feature code that sends me into that conference (*15 in my case). So far, a piece of

Hello! I need to use n-way call as it described here: http://habrahabr.ru/sandbox/52259/ It is in russian, but dial plan is quite clear. It works in asterisk 11: -- Blind transferring OOH323/7272-6385 to '0' (context fromtransfer) priority 1 -- Executing [0 at fromtransfer:1] NoOp("OOH323/7272-6385", "") in new stack -- Executing [0 at fromtransfer:1]

I'm trying to use Asterisk (v1.2.11) make a callback that dials both legs of the call into a Meetme() room together, but I keep getting "conf-invalid" messages. I created a callfile (/var/spool/asterisk/outgoing/out.call) that specifies a Local channel (extension) which contains a Dial() command to the "dialer", and an extension which contains a Dial() command to the

With 1.6.1, I'm trying to set up a test of meetme for creating dynamic conferences. extensions.conf: [meetme] exten => 2663,1,MeetMe(,De) 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 a PIN. Hangup. Then users would dial

[What it is] Web-MeetMe is a collection of PHP pages to allow for database driven scheduling of conference resources. [Current features] Schedule new conferences 1. Control start and end times 2. Set conference pin # a. Generate one if the requester leaves it blank b. Identify pin # conflicts (another conference with the same pin is scheduled at the same time) 3. Set Admin

I have a peculiar situation, and am hoping someone on the list can offer assistance. I am running CVS HEAD, and am using ITSPs for DIDs. The server has no Zap hardware, but is configured to use ztdummy. All incoming calls are via IAX2. Calls ring to SIP phones, voice mail, IVR, etc., with no trouble. I am also able to transfer calls among my SIP devices, voice mail, IVR, etc. All of my SIP

We are moving our asterisk 1.0 system to a new Asterisk @Home system (2.8) and I am the one in charge of doing it. I have run into a snag, though, on meetme conferences and with the transfer key. Regarding the transfer, it appears that both directions of all calls can transfer by pressing the # key. I do not like that ability. I would like to change it by doing 2 things: 1. Make the transfer

Work intrudes again and I will not be able to get to modifying the db and gui to support per-conference flags as soon as I expected. So I have released an update with what I do have available. [Location] http://www.fitawi.com/Asterisk [Features] 1. Schedule new conferences a. Control start and end times b. Set conference pin # i. Generate one if the requester leaves it blank ii.

Hello, So much keeps changing with the dialplan and Realtime lookups. Just downloaded the latest stable 1.6.1.2. The app_realtime, which was perfectly brilliant and did exactly what I needed, is gone; replaced with func_realtime. The REALTIME function is unacceptable: ; Get the conference number from the user exten => s,n(readconfno),Read(USER_CONFNO,conf-getconfno,0,3,20) ; See if

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

Software zur Erkennung von "Spam" auf dem Rechner priamus.teamware-gmbh.de hat die eingegangene E-mail als m?gliche "Spam"-Nachricht identifiziert. Die urspr?ngliche Nachricht wurde an diesen Bericht angeh?ngt, so dass Sie sie anschauen k?nnen (falls es doch eine legitime E-Mail ist) oder ?hnliche unerw?nschte Nachrichten in Zukunft markieren k?nnen. Bei Fragen zu diesem

Hi, I am using Group and Group_Count to limit the number of calls to go out over a single peer as our channels with that peer is limited to 8. If we dial and outside number over this peer and then transfer the call into a MeetMe conference the Group gets decremented when it should not? This is most likely an error on my behalf, however I am not sure what the correct solution is. I have set

Hi, I'm experimenting with the Meetme feature of Asterisk 1.2, exten => 2095,1,MeetMe(|Ds) This almost gives me what I want, where each employee can create their own on-the-fly conferences with a personal Conference Number and PIN. However, as the PIN is actually set by the first callee, then its subject to problems (first callee might enter the wrong PIN, and then no-one else can

We have two Linksys 942 phones which sound great when they call each other directly through Asterisk. But when they both dial in to a meetme conference room, the sound is very jittery. Other phones like Polycom 501 and Snom 360 sound fine when using meetme. Both Linksys phones are set to use the default g711u (ulaw) codecs. Adjusting the jitter buffer and jitter level settings to various values

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

Hello all ! How can I make conference authorization based on pin number ? I have: exten => 1,1,Meetme,1234|ps|2222 where 2222 is a pin number and this doesn't works Where do I have to add information about pin number ?? Greetings Andrzej Radke

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