similar to: meetme list concise

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

While researching Areski's new Web-MeetMe management gui, I found some odd (from what I expected) behaviour). Using the CLI to set un/mute status works but does not update the flags, or so it appears. Starting with a fresh conference (1 user) *CLI> meetme list 3456 User #: 1 Channel: OH323/R61 Using the CLI to mute the caller (no change in the user status0 *CLI> meetme mute 3456 1

This has to be the worst documentation I have ever come acrossed. I have found two or three docs on how to install it, but they are all so different and make huge assumption about what packages you have installed and locations of files. Has anyone seen something better, I want to get this working it is quite a cool app. Jordan Novak Communications Technician Logistics Health Inc. 1319 Saint

We've been holding back on this release to coincide with the Asterisk 1.4.0 release. This is mostly a compatibility release, but there are a few new features: * No longer requires register_globals in PHP * Separated code from configuration settings in ./lib/defines.php (hopefully this will make future upgrades easier) * Migrated all database interfaces to PEAR::DB which

[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

I thought I would give the new IMAP support a spin on my home server, but without much luck so far. Asterisk 1.4.0 Dovecot 0.99.14 Maildir format C-client 2006d The imap server is also the Asterisk server, so connections are on the localhost. The error posted to the logs is: IMAP Error: Can't open mailbox {127.0.0.1:143/imap/authuser=root//user=dan_austin}INBOX: invalid remote specification

Hi, I am using Debian OS kernel 2.6.22-3-amd64 and zaptel driver 1.4 with ztdummy module for meetme application. I use meetme with SIP channels. I have such problem that when one connects to the conference voice is "cut". Each voice sequence is disturbed. Does any one have similar issue and could give me some advice?? my extension.conf for meetme: ;switch =>

I'm afraid I am at a loss. If the three files, app_cbmysql.c, app_meetme2.c and Makefile all exist in ../apps then a patch -p1 from the ../asterisk directory should work. The -p1 tells patch to ignore the first directory in the path to the file in the patch, -p2 ignores two directories. Another option is to just edit the apps-meetme-cbmysql.txt and split it into three patchs and apply

I have V2.0.0 about read to roll. There's quite a few new features, clean-ups and bug fixes. I hope this will be the last major update for awhile, and as such I'd like to squash any issues before releasing it. New- ** Dynamic day/date listboxes to prevent invalid date entries ** Textbox focus and tabbing order ** CDR-like view at past conferences to show participants ** Vastly

I have a script that I want to leave running in the background to handle specific manager events. I'm running into a problem where it gets stuck in the wait_response function in phpagi-asmanager.php and the PHP maximum execute timeout kills the script. The script doesn't interact with the dialplan, so I cannot launch it from within Asterisk. Any pointers would be appreciated. I did

Does anyone know if it is possible to change the packetization time in Asterisk ? I was told by a client of mine that adjusting this with using G729 can greatly lower the amount of bandwidth used. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070819/b0cc470f/attachment.htm

Hello Everyone, We are currently experiencing some higher load on our servers, and since signaling comes into our servers on G729, we would like to implement G729 pass-through. A few questions arise, do we need to convert all the recording to the codec, and what about voicemail? We are also using A2Billing (hope I am not violating any thread rules), and would like to convert all that recording

Hi All; Can anyone advise if using Cisco IP Phones in skinny protocol is fine or not? Or it is better to use it in SIP protocol? Regards Bilal

Hello Team, I was looking for audio conferencing solution where i got Web-meetme. I had installed Asterisk 1.6.2.9 on Centos 5.4. Its perfecting working fine. I tried using Meetme even meetme app is working perfectly fine. I installed Webmeetme 4.0 and integrated with my asterisk. When i try to dial the conference number it take me to an IVR wherein it asks for the conference number. The time i

Sent from my Verizon Wireless 4G LTE DROID On Nov 19, 2016 10:26 AM, Sanjay Patel <spatel at rotateright.com<mailto:spatel at rotateright.com>> wrote: > > If I have my FMA intrinsics story straight now (thanks for the explanation, Hal!), I think it raises another question about IR canonicalization (and may affect the proposed revision to IR FMF): No, I think that we specifically

I tried using the iLBC codec, and whlie I like it, I ran into a strange issue. I did a few searches on Google and haven't found anyone with the same issue as this. Anyhow, I was using a Plantronics analog headset and box plugged into a Digium TDM card, dialed out over my VoIP provider's IAX channel to the PSTN. I was in a conference call which is running on an Avaya PBX (which

I have a Debian Sarge box that I am trying to get icecast2 and ices2 up and running on. I'm having problems, but do not know where to look to identify the problem. I have my audio working with ALSA - microphone in plays through my audio out. I'm running a basic configuration using the defaults provided, and the ALSA example config file for ices. When I try to connect to the server using

On Sat, Jun 27, 2015 at 2:16 PM, Nuno Lopes <nunoplopes at sapo.pt> wrote: > At least in C/C++ that's UB, yes. So you cannot map every turing machine to a valid C/C++ program then. :) Also, does this mean that "daemon" programs that run continuously till they're killed by the OS (using a mechanism that is not visible in C) are effectively undefined? -- Sanjoy >

The potential advantage I was considering would be more accurate cost modeling in the vectorizer, inliner, etc. Like min/max, this is another case where the sum of the IR parts is greater than the actual cost. Beyond that, it seems odd to me that we'd choose the longer IR expression of something that could be represented in a minimal form. I know we make practical concessions in IR based on

Hello, I am trying to apply the "called party identification" patch (patch 8824) and managed to make it work with a static data. Where do I take the name of the called person (the "equivalent" of CALLERID, but the other way...)? BTW, one note to the above patch: To make it work the device should have the parameter sendrpid set to true.