similar to: Meetme with video??? + $US 2,000 bounty

Hi Nicolas, There doesn't seem to be any interest in using asterisk and video. I posted a $1,000 bounty to get video meet me working without a single reply. I have now just bumped this to $2000 http://www.voip-info.org/tiki-index.php?page=Asterisk+bounty+Meet+Me+vid eo+conferencing This is a legitimate commercially binding bounty, I hope this might inspire some people to develop at least

I received this email link from Macromedia today http://www.macromedia.com/newsletters/edge/february2005/index.html?secti onIndex=3&trackingid=AWQX They are now selling Breeze video conferencing server by the minute as an asp, this service is powered by the macromedia communications server which is just a sip server like asterisk but for video. I'm still offering a $2,000 bounty

http://www.voip-info.org/tiki-index.php?page=Asterisk%20bounty%20Meet%20 Me%20video%20conferencing I posted this bounty for $US2,000 some months ago. Basically I needed the ability for 4 or 5 of us to conference on a weekly basis which is why I was happy to offer this bounty, however I have only had 2 people make brief inquiries and no one has really offered any substantial indication they

Ok people it's been a while since I've heard from anyone about this. http://www.voip-info.org/tiki-index.php?page=Asterisk%20bounty%20Meet%20 Me%20video%20conferencing The bounty for Asterisk Video Conferencing has now been bumped to $3,000. There are some additional feature and performance requirements for this funding now, please contact me if you are a developer who feel he has

I wonder if there is an application available, what would allow me to have a conference call (meetme) with video.

I have a doubt... is it posible to do Video Conferencing using asterisk? -- Facundo Ameal. fameal<at>gmail<dot>com Linux User #395088 Open your mind, use open source.

I am the one that placed the bounty. After it being there for 2 months and getting no takers (and very few if any people asking about it), we are almost finished writing it in house. I'll keep the bounty up untill we do finish our product so if anyone beats us to getting it working they'll get paid... W. Kevin Hunt CCIE #11841 MCSE, Linux+ SME www.huntbrothers.com >

Hello all, Just a quick note to let you all know that I have placed a bounty on Lauchpad to get bitrate peeling added to Vorbis. It is a feature that I think we would all like to have, and would probably pay something to get, but it hasn't been done. My request to you is to add to the bounty. I have seeded it with US$20, which is not enough to motivate a developer to get it done, but I am a

Hello, I would like to ask, if there's presence support in Asterisk and how to make it work with Xten's Eyebeam client. I tried searching all the possible documentation, google, but I found only a note, that there's a module in SER, that supports the feature. Is there also support in asterisk? Any pointer to documentation describing this is welcome. One more question -- is there

I was able to get it to work with 2 extensions. One for the "host" and one for the "participants" Not the best way to set it up but it works. Thanks -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Savoy, Kevin - Williston, ND Sent: Wednesday, December 13, 2006 8:06 AM To: Asterisk Users

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA256 Hello everybody, first of all I want to excuse myself for crossposting. I can't estimate which mailinglist could be the right one for our concern. Struktur AG, member of the Linux Solutions Group e.V., wants to announce an Open Source tender for an implementation of an audio codec compatible with Nellymoser Asao Codec. Struktur wants to pay

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA256 Hello everybody, first of all I want to excuse myself for crossposting. I can't estimate which mailinglist could be the right one for our concern. Struktur AG, member of the Linux Solutions Group e.V., wants to announce an Open Source tender for an implementation of an audio codec compatible with Nellymoser Asao Codec. Struktur wants to pay

Being bored to death by these long weekends with nothing to do? **** Why not go bounty-hunting? **** There are some feature requests in the bug tracker with monetary bounties attached. * Windows manager * FreeBSD Zaptel drivers http://bugs.digium.com/bug_view_page.php?bug_id=0000847 * IAX incoming/outgoing limit * 2B channel transfer on PRI * MGCP media gateway support All of these have

Is there a good Web Conferencing add-on or a compatible package for Asterisk? I know there are web based controls for the audio, but I am looking for PowerPoint or desktop sharing functionality similar to WebEx. Anyone using a package they like?

Just a trivial question: was the term "Meetme" invented for Asterisk as something like a brand name for its conferencing? Or was it an existing generic term for dial-in conferencing? Cheers Tony -- Tony Mountifield Work: tony@softins.co.uk - http://www.softins.co.uk Play: tony@mountifield.org - http://tony.mountifield.org

I sent a query about this before, but have some further information and am hoping somebody has a suggestion as to what to try next to debug this. I'm using an Asterisk box primarily for MeetMe conferencing. There are two sources: TDM via two Q.SIG T1's and SIP phones. Conferencing works fine between TDM channels. But when a SIP phone calls the conference, there's no voice path *to*

Hi All, I observed the following in my try towards Multiparty Conferencing. I am establishing the Multiparty Conferencing through Asterisk Manager API. I have two users SIP/111 and SIP/101 of which SIP/101 is treated as leader. Following commands are used - Action: Originate Channel: SIP/111 Application: MeetMe Data: |edwx ActionID: ffe4563 When I use the above, Incoming call will

I saw some of the heat about the $20 bounty earlier. So I don't want to put a low bounty out. Quote me a bounty, and I'll see if I can get it approved by management. :-) I'm in need of getting this bug fixed. Bug has all of the details, but basically 1.4.22 broke it all. I've waited as long as I can - hoping the bug would 'resolve itself' - but now I'm putting a

bounty offer prolonged to 31.4.2015 (end of april) Dne 3.3.2015 v 16:22 Marek Cervenka napsal(a): > hi, > > i'm offering bounty[1] $500 (five hundred) US dollars for resolving > https://issues.asterisk.org/jira/browse/ASTERISK-22708 > > fix must be available for asterisk 11.x and asterisk 13.x and accepted > to upstream > As part of this fix we should see seamless

I'm considering implementing an Asterisk PBX for conferencing. Before I get started, I wanted to make sure that it supports the features that I need. I plan to use Asterisk as a conference bridge only. I want people to be able to use my conference to listen live to lectures/etc, without having to listen to others in the conference. I'm using the FreePBX web interface, and I can't