similar to: Digitized audio at the beginning of a call

Dave wrote: > Hello, > I've got two streams, one for broadband, one for dialup. Well, having had > occation to use a dialup connection recently i checked the dialup stream. > Although it was streaming what the broadband stream was, the audio quality > was audibly worse. It didn't buffer, but it didn't sound as clear as the > broadband stream. This is expected.

I'm having a problem with a new SIP trunk. Calls within the UK to fixed lines are fine, but calls to mobiles have noticeably poorer audio quality. I thought it might have been a codec issue; we have used G.726 for internal and external calls (over primary ISDN and GSM). So I tried allowing "alaw", (G.711 A-law) which is the native codec used within the PSTN in this country,

Hi Jean-Marc, Sorry for late reply, thanks for interest. It's quality good for 10ms/audio, poorer for 20ms/audio. Quality equivalent for 10,20ms for mode=voip. PESQ was the tool that alerted me to something of interest, but I don't trust PESQ to almost any degree! It's good for hearing relative differences, of course, but not absolutes. Bitrate here was 28kbps, but I hear

Just attended the Cisco Networkers conference and sat through a presentation addressing in part the Cisco IP phones. All was presented openly without the need for disclosures, etc. The following is simply an FYI from the presentation, FWIW. Existing IP Phones (7905/7912/7940/7960): Transport: UDP Call Signalling: rfc2833 Security: Digest Auth Feature parity to SCCP: None Sip Enhanced

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I've have installed a new Asterisk 1.4.15 system after having previously used a 1.2 CVS head (from 10 Sep 2005). Both systems are pentiums though the newer one is actually a slower processor. On the new system, playback of gsm files is noticeably poorer (voice quality is flakely) on any connected phone (sip or isdn, internal or external).

hi All I need your help , for used Digium Card TE405P, for setting this Board AS E1 ISDN PRI. 1 .Current for make sure my config its rights or no I inform my configurations in Board Jumper T1/E1 is Closed is that rights or no ? for E1 i closed the Jumper. 2. I Want To seeting E1 in ASterisk/PC Back To Back To Cisco E1 AS5300 Use ISDN Signaling, my configutration :

I have a strange problem in that when I receive a call through the X100P which is forwarded to my budgetone 100 then the voice quality is perfect both directions. However, if I make a call out from the budgetone to the same caller via the X100P the sound level is a lot lower and the quality a lot poorer. I've had to set the rx tx gain to 1.5 or I can hardly hear at all. Any ideas what is

Hi, Running rsync with --whole-file yields poorer performance results than a simple "cp". Looking at the code - it looks like "copy_file" isn't really called when I add the --whole-file flag. The regular "receive_data" is doing the copy. Anyone encountered this? No other special flags (rsync --whole-file -av /mnt/ttt /home/sss/ ). Thanks, Amir --------------

List, I have a question about why plots sent to certain devices can be of so much worse quality than others. Whenever I plot to x11 or pdf the graphs are of excellent quality. eps is a bit poorer, but of passable quality, and jpeg or tiff are terrible. I tried all manner of parameters, on many versions of R (most major releases from about 2.2 - present) under linux and windows and on a few

Hi, > You can try the Ogg DirectShow filter to get Windows support for Speex: > http://www.illiminable.com/ogg/ if you wait a little longer, I am trying to provide speex support for the ffmpeg project (www.ffmpeg.org). It does encode/decode to/from avi and wav, a support for ogg has not been done, and my question is, should it be necessary? I don't really like ogg in first place..

On Mon, May 26, 2014 at 09:20:11PM +0200, Benno Schulenberg wrote: > > At the translationproject there are four more languages available > than are included in the 1.42.10 tarball: Danish, Esperanto, Malay, > and Ukrainian. Please include these in your next release. I deliberately didn't include some of these because the number of translated messages were pathetic. Danish has

Thanks for the reply. We are currently using AMBE (4Kbps) for our Traffic/Weather Channels. If you have ever had a chance to hear the service, you will know that AMBE does not do us well. I understand that 2Kbps is low quality, but any poorer than AMBE? If can get a decent quality for other low bandwidth talk channels, such as about 10-16Kbps and have it sound rather clean, then I would be

Greetings all- I have a CentOS 5 (Final) system that is serving up content to several other hosts via NFS. The amount of data transferred is rather small as most of the files are under 100kb and each export has maybe 100 files that are accessed regularly. I'm finding that as I add more hosts accessing the NFS server, the performance seems to be getting poorer. There are obvious delays when

Hi, (this is my first post here) A previous thread, starting Date: Tue 19 Nov 2002 - 06:09:56 EST "[vorbis] need speech and music in one" http://www.xiph.org/archives/vorbis/200211/0142.html expressed needs similar to mine, to encode a lengthy speech at low bitrate. I did some tests initially in September then concluded in December, and I was surprised to find Vorbis to be the best

I'm not sure if anyone is experienced with this but i'm having very strange issues with implementing speex in my iOS application. I have written a encode and decode function basically exactly like the example files online. To process audio I am using an iOS VoiceProcessingIO Audio Unit which is set at 16bit pcm with 8000 sample rate. After buffering 320 bytes of PCM audio I feed it into

Hello, I've got two streams, one for broadband, one for dialup. Well, having had occation to use a dialup connection recently i checked the dialup stream. Although it was streaming what the broadband stream was, the audio quality was audibly worse. It didn't buffer, but it didn't sound as clear as the broadband stream. I used lame to encode the tracks to mp3 and used it's

So I've got a bit of a head scratcher. Wanted to get some insight. I've got a PBX running 12.3.0 We're a ULAW shop from end to end. But I've been playing with G722 just for fun. I've got a Yealink T46G on my desk, And my colleague, A Polycom IP650 (Same office). Basically, Whenever I make an outbound call to a destination to something not G722 ready, I get no

Hi, I am currently using the aCGH package in R version 2.1.0 Windows with some supporting packages (eg. cluster) built under R 2.1.1.Using aCGH package, I am able to identify regions of genomic aberrations in my cell lines using the HMM model. However, when I tried to use aCGH for my paraffin embeded tumour sample, I got the following warning. Warning: MAD could not ben computed for one of

Anyone has tried wct4xxp drivers' alarmdebounce parameter ? I search the internet no one seems to have used it, is this how the parameter can be specified, eg :- # modprobe wct4xxp alarmdebounce=200 Does it have the effect of making asterisk PRI more "tolerant" with poorer quality lines ? Regards.

On Thu, May 29, 2014, at 8:33, Theodore Ts'o wrote: > It seems to me that if a message catalog has only 5% of the messages > translated, including them is very likely to result in a poorer user > experience than if everything was left in English. When it's just five percent, yes, that may be more of a nuisance than a help. But when it's ten to twenty percent, and when