similar to: Queue agent wrap up time.. .any ideas?

Hello, Does anyone know the developer(s) of the app_queue.so application? I'm looking for the PauseQueueMember and UnpauseQueueMember features of this application for the open source version that only seem to be available on the business edition of Asterisk. Thank You, Timothy Karl tkarl@imminc.com

Hello, We are about to attempt implementing realtime queues using Asterisk Business Edition and are wondering if anyone has had any experiences or recommendations /warnings with using this feature. One issue we're trying to decide on is if we should have a MySQL server running on the Asterisk server or a remote server to house the database/tables for the queue settings/members. The concerns

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:

Hello, Can anyone recommend a hardphone that has the following qualities... Both headset and handset ports Headset port has amplification built in Supports SIP using G729 We are switching from a Nortel switch to Asterisk. If anyone is familiar with Nortel phones, the Nortel 2216 phone has the features we're looking for in a SIP phone. Any help would be greatly appreciated. Thank you.

I am trying to find the app ChanSpy for Asterisk v1.0.7. I have tried looking on VOIP-info.org's ChanSpy page (http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+ChanSpy)and also referred to the link regarding bug 3836 (http://bugs.digium.com/bug_view_page.php?bug_id=0003836). I downloaded the attachments and tried to use the patch and compile the source. However, it seems that

Hi all, I'll try to explain my suggestion with my poor english. I've got the feeling that icecast is full of potential but not growing as fast as it can or simply not having the support it could have. The greatest lack are the "open doors" for new coders, webdesigners, translators, etc. Icecast is open: the sourcecode is open, the mailing list is open, etc. But a skill is

When I installed my first home-PBX three years ago, I was looking at "cellsockets" -- devices which will accept certain cellular phones and provide an RJ11 jack, generating the ring-voltage and recognizing DTMF, which in turn makes your cell-phone look like a CO line. Pretty cool stuff, in theory, but it just didn't seem to be worth the cost, especially since it locks you to a

Hello together after trying to deploy one of the latest version out of the KH branch to our servers we ran into an issue with streaming of ondemand files. After deploying 2.3.2-kh27.1 and also trying 2.3.2-kh24 and 2.3.2-kh22 we see that if we try to stream an ondemand file through the fileserv option we get the ondemand file play back correctly and also see the appropriate file length in

I have the following helper method in application_helper.rb: def format_date(date) day = to_s(date.day) month = to_s(date.month) time = to_s(date.time) date = day + "/" + month + " - " + time return date end I am trying to call this method in a view like this: <%= format_date(bounty.created_on) %> create_on is a timestamp in mysql. I am

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

> On Sat, 2005-03-05 at 23:52, Mihail Egorov wrote: > > 1. How can I identify disconnect due to low <queue-size>. Suppose, I have > > enabled loglevel=4 (debug). Suppose, I have network jam. What shall I see at > > error.log? > > There is a log message that signifies the removal of a listener for > being too slow and that is > "Client has fallen too far

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'm trying to find out if Asterisk will support plain G729 or G729b. I've read all over that it supports G729, but I can't seem to find any explicit remarks regarding the specific versions of the codec Asterisk will support. I noticed that Digium allows Asterisk users to register and download G729a, but refers to it as G729 on it's pages. I also noticed that on

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

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

I don't know if anyone noticed my post a few months ago on the asterisk-user mailing list, but I've been writing a queue log parser. I was wondering if anyone had any queue_logs (the bigger the better) that I would use as sample data? I would of course be willing to post the stats up for the people who send me their sample logs. Post them to me off list, and please tell me if you are

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

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 >

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

hi, there is T.38 fax gateway for asterisk http://bugs.digium.com/view.php?id=12931 please test it and report bugs for people from http://www.voip-info.org/wiki-Asterisk+T.38+Bounty if you still want donate t.38 development please contact me at cervajs at fpf.slu.cz --------------------------------------- Marek Cervenka =======================================