search for: stien

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2005 Jan 03
1
realtime audio for asterisk using jack
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
2005 Jan 03
1
echo test application delay using the asterisk cli
...here does the latency stem from?. Asking this on the asterisk irc channel gets me nowhere. I've tried many many times formulating it different ways, but few seems to understand I'm only using the interactive shell mode of asterisk dialing the echo application on the same server. -- Esben Stien is b0ef@esben-stien.name http://www.esben-stien.name irc://irc.esben-stien.name/%23contact [sip|iax]:b0ef@esben-stien.name
2008 Jun 28
1
Missing Window Border
...orthy in the xorg log. I have a Radeon 9250 (rv280) card and I run with the DRI drivers, ati, xf86-video-ati-6.8.191. I run Mesa-7.0.3 and xorg-server-1.4.2 on GNU/Linux-2.6.26-rc8. I'm running all the compiz packages from the 0.7.6 directory. Any pointers as to what I can try?. -- Esben Stien is b0ef at e s a http://www. s t n m irc://irc. b - i . e/%23contact sip:b0ef@ e e jid:b0ef@ n n
2005 Aug 11
1
Firewall will definately increasejittersinyourvoice conversation
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" <jonathan@bluegrass.net> writes: > even when I have taken all other security measures...
2005 May 08
2
SPEEX LADSPA Plugin
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
2005 Jun 13
1
Re: Re: Digium Website Update: Asterisk Busi ness Edition
> -----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'...
2013 Jul 15
0
[LLVMdev] libcompiler_rt.a, No such file or directory
...ler_rt.a»cp: cannot stat «/pkg/llvm-3.3.src/tools/clang/runtime/compiler-rt/clang_linux/san-i386/libcompiler_rt.a»: Ingen slik fil eller filkatalog : Ingen slik fil eller filkatalog I'm using gcc-4.8.1 and glibc-2.17 and I'm on 32 bit GNU/Linux. Any idea as to what I can try?. -- Esben Stien is b0ef at e s a http://www. s t n m irc://irc. b - i . e/%23contact sip:b0ef@ e e jid:b0ef@ n n
2005 Apr 14
2
pre-processing for audio quality
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.
2005 May 08
0
Heavy CPU Usage During SPEEX Calls
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 [sip|iax]: e e jid:b0ef@ n n
2005 May 18
0
Missing Transfer Command (asterisk CVS 20050518)
...mmy 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 b0ef@e s a http://www. s t n m irc://irc. b - i . e/%23contact [sip|iax]: e e jid:b0ef@ n n
2005 Aug 09
2
Asterisk and Wave files problem
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
2005 Aug 10
0
Asterisk Stops Sending Data (CVS 20050809)
...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 weeks. Anyone experienced this? -- 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
2005 Aug 11
2
Sip ports
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
2007 Jan 03
0
Root Visual not a Double Buffered GL Visual (compiz-GIT-20061223)
...irect rendering is working fine. I've seen numerous people mention this problem, but they seem to be using proprietary drivers. My xorg log shows: (**) Option "AIGLX" "on" (**) Extension "Composite" is enabled Any pointers as to what may cause this? -- Esben Stien is b0ef@e s a http://www. s t n m irc://irc. b - i . e/%23contact sip:b0ef@ e e jid:b0ef@ n n
2005 Jun 12
1
Comparison
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