similar to: realtime audio for asterisk using jack

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"

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

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

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

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

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

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

i have added port=5060 to sip client configuration but it seems the same problem and in the same errors: Aug 11 10:29:18 WARNING[9869]: chan_sip.c:843 retrans_pkt: Maximum retries exceeded on call 04b3ccd87e45e719588c54a4017e3b99@172.16.180.21 for seqno 102 (Non-critical Response) __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam

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.

Im haveing a problem with my profiles share on my Samba 2.2.3 PDC server. I have a share like this: [profiles] path = /home/samba/profiles writeable = yes create mask = 0700 directory mask = 0700 browsable = no valid users = root,@smbusers The roaming profile works just fine with windows2k, and the users can't read the other profiles (they get a "access

Hi Guys, I'd reflect Vieri's question actually, because there must be a way of doing it? Maybe not with ICES, but maybe combines with JackIT? The reason for my interest is I work in a radio station in Spain, and because of the geographics where I am we have to relay around a lot. If there is a way of sending an incoming feed into ICES, and sending it out in various directions, would

Hello Im from Denmark and i've just got my Digium TE405P. But i have some problems when i connect it to my E1 connection (ISDN30). My telco delivered a alcatel box witch have a G.703 120ohm (DB9 with a serial to rj45) and a 75ohm coax connection. I've tried to connect using the 120ohm with rj45 and a ordinary utp cable. But it dosent seem to work. I've tried several zaptel.conf

Karl Heyes <karl@xiph.org> writes: > Are you running with realtime privileges, for this you need to start as > root if you want that. Even with realtime privileges there may be odd > cases where the scheduling latency is a bit too high, it all depends on > the drivers and kernel version but the current state is not that bad and > getting better. There are several ways to

Hi, Question: Can a TCP server (running on the same host as the OpenSSH server) know the user id/name of a user forwarding an TCP port ? I.e. if someone on some client machine does ssh -L9999:localhost:9999 someuser at somehost nc localhost 9999 and a service accepts the connection on port localhost:9999 on somehost, can it somehow safely read out the user name "someuser"? Long

Daniel Webb wrote: >>From what I have read on some mailing lists, there is a jack module forices2 > that works quite well. Why isn't this information on the ices2 web page or > documentation? Ices is useless to me without this, yet so awesome withit. I > saw requests from people for you to do this over a month ago. What's the > problem? talk about coincidene: