similar to: conference question...

Hello all, I'm in final testing stages and preparing training for a new Asterisk rollout. I'm replacing a Cisco Call Manager system, and re-flashing the 79x1 phones with SIP 8.5.2. With the SIP load and while in a call, I use the "Confrn" softkey to invite other participants. I can add one other participant endpoint into the conference, but no more. I know I can (and

Hello, using Asterisk 1.8.12.2 I am having trouble with exiting the conference room by entering a single digit. option X of the Meetme()-application should do this. I have following in extensions.conf : /exten => _1000X,n,Set(MEETME_EXIT_CONTEXT=dynamic-nway-invite)// //exten => _1000X,n,MeetMe(${CONFNO},dMX)// // // //[dynamic-nway-invite]// //exten => 0,1,NoOp(confno =

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? *

Is it possible to use an Asterisk feature code to transfer a call to a specific extension? For instance, if you take a call, and the caller wants to go to a conference, it would be nice to use a feature code for this, rather than going through a longer transfer sequence. e.g.: - You have a meetme conference: [conferences] exten => 21,1,NoOp(MeetMe Conference) exten => 21,n,MeetMe(50,pM)

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 - We have a couple of large spaces that we'd like to cover with dedicated conference units like the Polycom Soundstation IP3000. We're concerned about adequately covering the spaces, though, one of which is very long and narrow. I wanted to get external add-on microphones for the IP3000, but I've found that unlike some of their other conference products, it does not have this

I'm having a problem with conferences both meetme and app_conference, though I've done most of the testing with meetme. Essentially, I get little gaps in the audio - usually fewer than a dozen or so samples, though it does vary. They seem to occur at random, but I usually get one ever few seconds, on average. They also seem to delay some buffer somewhere, so that if I start recording

Hey! I am trying to configure the voice onference with MeetMe application for my internal users. I have my server and 4 clients on same LAN and following is my extensions.conf file: [globals] Ahsen=SIP/222 Tahami=SIP/444 Uzair=SIP/333 Wasif=SIP/555 [internal] exten => 1234,1,Macro(voicemail,${Ahsen}) exten => 4321,1,Macro(voicemail,${Uzair}) exten => 5678,1,Macro(voicemail,${Tahami})

Hello; We need to have admin page, so the administrator can create passwords to be used to join the meetme conferences and can determine the allowed time ..? Well, the admin interface can be done easy (I do not know if there is something ready), and the password and the time limitation can be added to the database (or even text file), but how asterisk can use it? Do I need to use the AGI to

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

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 "user1" calls user "user2" 2. "user1" then presses the feature code "*0" to redirect "user2" to conference room 300 3. "user1" then dials the user "user3" 4.

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

[New Features] 1. Added focus and tab-order to all input fields 2. Dynamic generation of date/month/year listboxes a. It is no longer possible to schedule an invalid date. 3. Added 'Extend' and 'End Now' buttons to the monitor page. 4. Invite button on the monitor page. This greatly simplifies the process of adding callers to a conference. The ./lib/defines

Conference Invitation "Science and Education" foundation and its partners kindly invite you to take part in the scientific events that we are holding in Elenite Holiday Village, Bulgaria during 2014. All the papers presented will be published in open access online journals, part of International Scientific Publications (www.scientific-publications.net). Conferences in June 2014 The

Conference Invitation "Science and Education" foundation and its partners kindly invite you to take part in the scientific events that we are holding in Elenite Holiday Village, Bulgaria during 2014. All the papers presented will be published in open access online journals, part of International Scientific Publications (www.scientific-publications.net). Conferences in June 2014 The

Conference Invitation "Science and Education" foundation and its partners kindly invite you to take part in the scientific events that we are holding in Elenite Holiday Village, Bulgaria during 2014. All the papers presented will be published in open access online journals, part of International Scientific Publications (www.scientific-publications.net). Conferences in June 2014 The

Hello Users, Is Without Zaptel interface Installed, conference Bridge is worked or not. Why it need, For SIP conferences through OpenSER.... Please Help me For me its Giving Some Errors and warnings. *== Parsing '/etc/asterisk/meetme.conf': Found Oct 25 18:16:13 WARNING[12281]: chan_zap.c:913 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory Oct 25

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

Hi. I have a problem with my audio in meetme conference under asterisk 13 using Debian buster compiled from source. The problem is that there is some sort of distortion in the audio -- a workaround is always to lower the listen volume (*4). I see nothing in the log and so I wonder what is happening. I have dahdi loaded so I can record the conferences. Thanks in advance for any suggestions and

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