similar to: Sip ports

No matter what I do, I'm not able to get window borders working. Both commands "emerald --replace" and "gtk-window-decorator --replace" runs fine, with no output, but no window borders are drawn. I've tried with and without: Option "AddARGBGLXVisuals" "True" ..defined in the Screen section of xorg.conf, but with no difference. To my

No argument here..... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Esben Stien Sent: Thursday, August 11, 2005 8:11 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Firewall will definately increasejittersinyourvoice conversation "Jonathan k. Creasy"

Is there a ladspa speex plugin available or is anyone working on such a plugin?. -- Esben Stien is b0ef@e s a http://www. s t n m irc://irc. b - i . e/%23contact [sip|iax]: e e jid:b0ef@ n n

> -----Original Message----- > From: Esben Stien [mailto:b0ef@esben-stien.name] > The other problem is the issue that free software developers are > mostly (in my experience) not happy with the fact that their code > would be used in proprietary software. It conflicts with the whole > religion of free software. Well, yeah, that's the whole problem, isn't it? You

Hi, I'm recording wave files but I cant get Asterisk to play them, only if they are in 8000 Hz. What is the maximum sample rate Asterisk can handle? I have been using 16-bit 44.1, 22050 and finally 8000 kHz. Many thanks, Christian

We are using Speex as our major codec for voice application. We like the Speex solution so far. Currently, We have tried to compare voice quality between different public available VOIP solutions such as Skype and others. We notice Skype use higher CPU for signal processing. Is it because of this extra work the audio quality sounds clear? (Sometimes, it sounds like the audio signal is not real.

i have been working on a voip client that goes head-to-head with skype in technological terms. for this, we used speex wide-band codec. without the denoiser or the pre-processor, i find that speex quality at 16 khz sampling, 16-bit samples (mono) to be clearly superior to anything that skype offers. even though, at the moment, i am not using packet loss compensation, i find that speex is

Any plans for asterisk to support jack for realtime audio?, http://jackit.sf.net -- Esben Stien is b0ef@esben-stien.name http://www.esben-stien.name irc://irc.esben-stien.name/%23contact [sip|iax]:b0ef@esben-stien.name

I'm trying to figure out why I get delay using the echo test application. I'm using the asterisk cli so I got no external factors that could interfere. I'm getting close to half a second delay speaking into the microphone and hearing it out through my speakers. I'm doing lots of audio work (sequencing and recording) so asterisk is definitely the problem. Of course, only thing

when i send calls from an asterisk box to a voip provider the call fails and give me these messages: *CLI> Sep 23 21:32:32 WARNING[14595]: chan_sip.c:6890 handle_response: Forbidden - wrong password on authentication for INVITE to '"asterisk" <sip:asterisk@195.112.214.99:5070>;tag=as19e688a1' -- SIP/call-0f60 is circuit-busy == Everyone is busy/congested at this

It does not look like Nufone is still in business, judging from the content on their site, which is very little. There is not even a configuration document to download, to connect to their network. The rates file is only for US/Canada calling. No international rates on this rates.csv file. I have signed up with a $5.00 account with them way back in November 2004. After signup, I havent received

Hi: Iam looking for a sip softphone that supports g729 codec Any one have an idea ? Reagrds; jonnyhashem --------------------------------- Don't get soaked. Take a quick peak at the forecast with theYahoo! Search weather shortcut. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Trying to run compiz on xorg-7.1 (aiglx), but getting the message: compiz: Root visual is not a double buffered GL visual compiz: Failed to manage screen: 0 compiz: No manageable screens found on display :0.0 I'm on GNU/Linux with kernel 2.6.18 using DRI on ATI Radeon 9250 (rv280). I have mesa-6.5. Direct rendering is working fine. I've seen numerous people mention this problem, but

Trying to compile llvm-3.3 and I get this: llvm[4]: Copying runtime library linux/asan-i386 to build dir cp: cannot stat «/pkg/llvm-3.3.src/tools/clang/runtime/compiler-rt/clang_linux/full-i386/libcompiler_rt.a»: Ingen slik fil eller filkatalog llvm[4]: Copying runtime library linux/ubsan-i386 to build dir llvm[4]: Copying runtime library linux/ubsan_cxx-i386 to build dir make[4]: ***

I'm getting close to 90% CPU usage when doing SPEEX calls. When using GSM everything is fine. This has only happened in the last months with CVS HEAD. I'm running now CVS as of yesterday on linux-2.6.12-rc3-RT-V0.7.46-02. Anyone else experienced this? -- Esben Stien is b0ef@e s a http://www. s t n m irc://irc. b - i . e/%23contact

What happened to the transfer command?; I can't find it in recent CVS. I got ztdummy and zaptel loaded on a linux-2.6.12-rc3-RT-V0.7.46-02 and I'm able to dial into meetme, but I can't find the transfer command to transfer a call from the asterisk cli. Is this function removed and only found in the manager interface or is there something wrong with my setup? -- Esben Stien is

Been having problems with CVS lately. I get incoming calls from an iaxcomm user, using a windows system. Asterisk stops sending data after about 30 seconds. I view this with tcpdump on the same computer. I still receive data and can hear the remote party. This problem starting sometime mid last month. I regularly build cvs and have run into this issue before. Now it's been like this for some

How suitable do the developers think that LLVM would be as a code-generator for a dynamically typed langage? Would the lack of static type information make the traditional code optimizations performed by LLVM relatively ineffective? Sincerely, Rayiner Hashem

HI: what is command on console to know how many calls are sending in the same time? __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

> That said, many dynamic languages should be cleanly mappable to the LLVM > layer. What are you thinking about in particular? Not thinking of one in particular right now, but it would probably something like an object-oriented Scheme. > recommend writing some language tuned optimizations (as necessary) that > would lower the dynamic objects into the primitives the optimizers >