similar to: SMDI Bounty - where?

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040331/c2abf19f/attachment.htm -------------- next part -------------- Hello, Is there any work in progress for supporting SMDI in Asterisk ? if Not, could anyone tell how to get started implementing it for Asterisk. Regards, Tony

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

Hi, I've just initiated a new bounty for the above; http://www.voip-info.org/wiki-Asterisk+bounty+callingcard+to+MySQL Any takers or any contributors please respond to me privately. I do not know exactly how the bounty process works, but I can coordinate on this ? SW -------------- next part -------------- An HTML attachment was scrubbed... URL:

The bounty stands at $5,500. I'm seriously considering taking a shot at it if I can find a decent T.38 provider to test with (I'm still hoping for reliable PAYG T.38). It looks like a lot of very smart people have done a lot of very hard work (t38modem, spandsp) that would go towards getting this working. At this point it appears to be mostly a matter of integration

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

The CDW part number is: 672205 And the cisco part number is: CON-SNT-CP7960 Hope this helps... In fact I think I will add it to the Wiki. ~c ------------------------------ Message: 11 Date: Mon, 27 Sep 2004 10:40:38 -0500 From: "W. Kevin Hunt" <Kevin@hbcorporate.com> Subject: RE: Cisco Downloads --> was --> Re: [Asterisk-Users] Cisco 7960 andAsterisk...not working... To:

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

http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+simultaneous+registry From the WIKI: Contributions Manager: Daniel Jimenez (cuban) Bounty: $50 USD Date opened: July 10, 2004 Contributors: cuban ($50) Detail Yes, Yes I know you could do all sorts of fun with the dialplan to produce a similar effect, but I still would like to be able to do this. Plus it's easy money :). I

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

I've just received an email from a colleague who told me to put my money where my mouth is .... So here it is - I'm offering $500 and looking for other people to add to this bounty. We can get a group of people putting matching funds up to finalize the scope of the first Open Source Asterisk / Wave conference call integration robot bounty but if you have any other suggestions feel free

Wrong list. asterisk-dev is for changing the C source code of Asterisk. I don't think AGI's "count" or are considered for inclusion into the subversion repository as stated by one of your conditions for payment. On Wed, 29 Apr 2009, Alistair Cunningham wrote: > I'd like to offer a bounty for a feature for Asterisk where an AGI > program can park and retrieve calls

(forgive the brief interruption to -users with a mostly -dev issue, just wanted to publicize this on behalf of the larger community) If there are any ambitious coders out there (not too many shekels yet, but I expect some folks may pony-up) please see: www.voip-info.org/?page=Asterisk+Bounty+Email+TTS We are at $150 & counting. Maybe lobby your exec's for $50 to contribute to this,

I'm still unable to get my SPA-1001 to work behind NAT with an Asterisk box out on the Internet. It works fine to my local A@H box. I've tried... many things. I'm willing to pay a $250 bounty (PayPal preferred) to anyone who can help me get it working. Any Sipura experts out there? Eric.

http://saunderslog.com/2009/03/03/voxeo-launches-tropocom-mashup-platfor m/ I'll pay anyone a $20 bounty for someone to replicate the USA Asterisk Weather App on Tropo. Would like to see how quickly this is implemented. Regards, Dean Collins Cognation Inc dean at cognation.net <mailto:dean at cognation.net> +1-212-203-4357 New York +61-2-9016-5642 (Sydney

Hello: I've been wrestling with an integration issue for which there also seems to be one piece missing. I'm hoping someone on this list can help. Due to a complete lack of cooperation from our current voice provider we are in need of an alternative way to access our Octel 350 voicemail system from our SIP (SER) proxy. I can redirect and relay calls to numerous destinations via

We have recently pulled an ancient Fujitsu-branded Centigram voicemail system out of production use and replaced it with an Asterisk box, which is now serving as our enterprise voicemail system and automated attendant. The Asterisk system is connected to a Fujitsu F9600 PBX and uses the 1.6.1.x SMDI module to communicate between the two systems. Save for one issue, the replacement of the old

Hi all, I recently purchased a Junghanns ISDNguard and to my horror I found out: - Junghanns technical support is non-existant - I can't use it without recompiling Asterisk with res_watchdog My situation: - Recompiling Asterisk with thrid party code is not an option us - We only need it for manual failover - don't require all the fancy monitoring stuff I need a simple shell script

I'm in process of transitioning a number of offices to a hosted virtual pbx from Junction Networks. It's a combination of OpenSER and Asterisk. They have a nice click-to-call extension for Firefox, but I need the equivalent for IE so that it can work with our CRM system. Junction told me that they have a bounty on offer for this if someone's interested in doing the work. Would the

Steve Blair writes > I can redirect and relay calls to numerous destinations via >SER but because the Octel needs an SMDI interface for mailbox >identification I am stuck, none of the solutions thus far support >SMDI-SIP munging. > > I just started thinking about the possibility of using Asterisk >with a few FXS cards to provide the gateway between SIP and >the Octel.

can you please share the cdw part # for the $ 10 service contract ? W. Kevin Hunt CCIE #11841 MCSE, Linux+ SME www.huntbrothers.com > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Christopher Jacob > Sent: Saturday, September 25, 2004 9:51 PM > To: