search for: gladdening

http://bugs.freedesktop.org/show_bug.cgi?id=15494 Summary: http://www.gladdening.com/games/mariorush/mariorush.html does not render/play correctly Product: swfdec Version: unspecified Platform: Other OS/Version: All Status: NEW Severity: normal Priority: medium Component: plugin...

The scenario we have is fax send/recieve software that ONLY talks T38 and an asterisk box. We have ITSP providers that do NOT talk T38 but G711 only. Does asterisk have the capability to take the T38 call from an ATA or T38 software then bridge/transcode it and do G711 out to the PSTN providers? If not is there another product PAID or FREE software or hardware that can do this easily and

I have a small pile of them from customers who were too lazy to send them back after switching to our local voice service... Is there any hope of ever using these things with Asterisk? Vonage does not want them back and they won't unlock them either. A terrible shame! Should I just toss them? Steve

I'm looking for some help in how to keep asterisk from doing this. If we loose Internet or routing to our upstream provider even for only a few short minutes asterisk quickly gives up & never tries again. I have to do a manual reload to get it to register with my sip provider(s) again before incoming calls are accepted. This is really bad as it causes us to loose the ability to get

I keep running into the dead end that it can't find config.h in the source tree. It looks like newer kernels don't use it anymore. Someone ran into this awhile back when compiling 1.2 and it looks as though the issue was never resolved. Any ideas? Last time I tried this, I was on fedora core 5 64bit and all went well. It's not working on this newer setup Any ideas on what I can do

been testing with a rather simple setup. The mission is to actually get a reinvite to work on the lan. I am trying with two sipura phones G.711 codec forced on both both on the lan no nat no fancy options suchs as tT or H No matter what we do asterisk hangs on to the media path, how in the world do I get a reinvite to work where the media path is actually handled by the two phones on the lan?

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Still looking for some direction with this subject: I think the term is called multi-line appearance.... Is this something that is directly supported in Asterisk? I can't seem to find any information on it or how to actually use it.... This is where you have several sipura-841 SIP phones for example... if someone pickes up 'line1' I'd like the light to come on on ALL phones to

Yes, thank you, specifically and all. On this platform, we call what I will use a "frame stack" (it's actually not a stack, but that's really not relevant). Special thanks to Mr. Mendell--I promise to make good use of the contents of lib/Transforms/Utils/PromoteMemoryToRegister.cpp. To All, I'm sorry I wasn't clear in my original posting. In hindsight, it was clearly

Sorry this is a bit of a newbie question, I've been at this for a few months and still have not quite figured this one out. I've been able to setup one itsp (incoming calls) (sip account) with a register line like this: register => nnnnnnn:ppppp@sip.provider.net -or- register => nnnnnnn:ppppp@sip.provider.net/nnn to come directly into an extension in the dialplan It seems that

http://bugs.freedesktop.org/show_bug.cgi?id=15601 Summary: Mario Rush hero falls through the floor - collision detection error Product: swfdec Version: 0.5.5 Platform: All URL: http://www.gladdening.com/games/mariorush/mariorusharena .swf OS/Version: Linux (All) Status: NEW Severity: normal Priority: medium Component: library AssignedTo: swfdec at lists.freedesktop.org ReportedBy: astralstorm at gmail.com...

On the same P2 450Mhz box..... I have tried both UHCI usb on a 2.4 kernel and enhanced RTC on a 2.6 kernel. Have not tried UHCI USB on a 2.6 kernel as of yet. Both seem to work GREAT. I have read in many places to be sure to use a digium card as a time source and not to reply on the cheap solutions. However I have regular meetme sessions of 5 and 6 people at the same time that frequently go on

This was just a recent personal experience.... Maybe I missed a thread on this: We recently installed asterisk (CVS-HEAD) on a debian system using 2.6 kernel and the enhanced RTC for all timing. Also a custom compiled kernel for the CPU on the box (P4). We had a strange thing happen in that with Debian's MPG123 package: Sound files played in asterisk/mpg123 were heard at literally 1/10th

I've recently installed the latest Debian Linux for powerpc onto and old iMac (version A) the original iMac with a 233Mhz G3 processor and 160MB of sdram. The debian install went smooth and so the the apt-get install of Asterisk 1.4.21 It appears to have no functioning zaptel or ztdummy module. Is because of hardware? or is it because whoever built the package didn't include a full

I am using asterisk (2 week old CVS) am for the first time have been starting to experiment with busy and congestion. At this point I am only using sip endpoints PAP2-NA devices. All testing of this is being done on a local network. my test extension looks like this: exten => 7777,1,Answer exten => 7777,2,busy(35) exten => 7777,3,Hangup Or like this: exten => 7777,1,Answer

I'm going to try and ask this again and keep it short and as too the point as I can while still providing enough info to be of use. PLEASE advise if I am going about this wrong or asking too much. I'm seriously doing my BEST to throughly read the docs and try a bunch of things BEFORE coming here to ask and possibly annoy. If is documentation that explains thsi process in terms that

Scratching my head and trying this. Asterisk Version: Asterisk 1.4.21.2 Tried: exten => 4771,1,ExtenSpy(4724 at mbb) exten => 4771,2,Hangup Also tried: exten => 4771,1,Answer exten => 4771,2,ExtenSpy(4724 at mbb) exten => 4771,3,Hangup Also tried many variations including option ,b I think most calls I make are 'bridged' extensions 4771 and 4724 are both in mbb context.

Trying to compile asterisk (again) from scratch. I seem to be still experiencing the effects fro Jan 25 where I get no sip to sip audio. I have tried upgrading to 1.2.3 which has made no change in the problem. I am starting over and now trying to compile/install /trunk zaptel libpri asterisk following the instructions to grab the source trees: # svn checkout

Been trying to get this to go but nongo :-). I'm asking for some guidance especially if I should not be doing this on an RT kernel. I've installed what is supposed to be all of the requred deps. Some factors that may be adding to my problem are: 1. this is only a test.. it's a 32bit guest OS running in VMware under a 64 bit windows host.. (although I'be compiled it on other

New to Aserisk 1.6 and find the 'installation tutorials' seem low to non existent. You go to the main Asterisk page (digium.org) and really just old install instructions for 1.2 are in the examples. Download links only give you asterisk itself and not dahdi or libpri which also are needed to run asterisk? It's very confusing to anyone who is new. Someone take notice! we need a link