similar to: Weird noise problem on SIP transfers...

Using narrow, wideband, and ultra-wideband encoding on a short 16khz wav gave .spx's of 3,789 ... 2,935 ... and 1,875 bytes. Even after reading the manual, smaller files for the higher frequency encoding seems counter-intuitive. My mp3 at 32 kbps on the original 22khz wav is 3,866 with a quality comparable to speex wideband on the converted 16khz wav, so speex is a 24% improvement in size.

Hello all We have tested for a mounth or two an asterisk PBX using one T1 channel bank with 24 fxs and one TDM400P digium card with 4 FXO modules. This worked with minor problems, the most notorious being some sporadic static noice or failure in the first FXO module on the wildcard. Now we have a client with 12 pstn lines and 48 extensions and we are trying to deploy an Asterisk PBX server

Hi i've read this post http://lists.digium.com/pipermail/asterisk-dev/2007-May/027666.html I just want to know if there are some upgrades... on 1.4 or 1.2. I'd like to store two records in the CDR instead of one, when a call is transferd. Is it possibile now? Thanks to all -- /*************/ nik600 https://sourceforge.net/projects/ccmanager

Hello, I use a Suse 11.0 as a Samba 3.2 PDC. The clients run XP SP3. I have upgraded a few weeks ago from Suse 10.3 and now all files tranfer that I do - for example, downloading a file using a web browser - it leaves a trash file named "transferd-file:Zone.Identifier" or "tranferd-file:encryptable". The odd thing is that from Windows I can`t read/write/delete these files.

Hi How do i set up so that everyone can dial, for example *21* to forward all calls to a cellphone or another extension and how do I enable so that cals can be transferd between extentions. I use asterisk 1.6 and have my phones in unistim.conf and my extensions in extensions.conf. Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Odengatan 106, 113

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi to everybody. First of all I hope this is the right place to discuss such an (nitpicky) issue. I've just been testing the current Opus release and for mere curiosity compared its performance to WMAPro with CD quality music at low bitrates (48 kbit/s). While Opus generally does a very good job, I found one particular example (a high pitched

Hi Everybody I've got a weird problem. I am running one Asterisk system on a dual processor box. This box mostly do VoIP only but it has a Fritz PCI ISDN card installed with latest drivers. Dialing out through the ISDN cards from an internal Snom phone works fine and so does dialing in. Except - if I load the ztdummy module (for IAX channels) the capi drivers starts acting up. It is hard

I'm running a X100P connected to a POTS line and a TDMP400P w/ two FXS daughter cards. Both calling out from one of the FXS phones (internally) or calling my home number (externally) the FXO card starts to freak out. By freak out I mean I can still hear but it sounds like you are underwater, there is an annoying hiss or buzz on the line as well. If I hang up and pick up another house phone

I was doing some testing with RC2 and I noticed that RC2 doesn't encode past 19kHz with this clip (-b256 and -b350). There are no problems with this clip like it was before, but this clip contains signal past 19kHz which is audible as a faint high-frequency hiss - and that hiss is gone in the encoded file since RC2 cuts off at 19kHz. I think that -b256 and -b350 should encode at least up to

i have been using the asterisk for some three weeks. Previously i was using the softphone iax-phone and now i have to shift to the sip phone xlite. The problem is that there's always unbearable noice in sip to sip calls. Is there any way to get rid of this???? Kindest MM Luqman -------------- next part -------------- An HTML attachment was scrubbed... URL:

How can I add some size to SPEEX encoded packets without affecting decoding results? I need it to fit smaller (due to VAD) packets in CBR acm-wav file. i.e. I have 10 bytes packet with 0.02s of silence (only background noice without speech). I need to fit it in 16kbps CBR wav file, so I need to put 30 additional bytes to this packets, but decoder should still decode only 0.02s of background

hi All, We are using Asterisk server with sip phones (SJPhone). On the local LAN, when we use the SJPhone as the SIP client, communication works fine with no disturbances and noices. But when it comes to dialup connection we harldy hear anything except a rough noice. We have included G.729 Codec (Annex B) with the Asterisk server, and we added the G.729 Codec to the SJPhone too. But it seems

Hi, I have got a hissing/humming noise on my handset that is connected to the S100U USB device.. I was running a CVS version of * that was about 2 to 3 weeks old and had perfect quality.. But that HDD died on me so I reinstalled and used the latest CVS, now I have this sound that is best describes as a similar sound to the noise a cpu fan makes.. Like a high piched hissing noise.. Any one

hello all- has anyone had any problems with the handsets on BT100's. Just picked one up for my lab and the speakerphone works great but I am only getting one way audio (incoming) from the handset. Since the speakerphone works fine, I can't think of any config. reasons why the handset wouldn't other than a faulty handset. Any thoughts or experiences? Jason Kawakami Technical

Hi, I use shorewall on 2 computers and I''m really happy with it. But now I came over a special case where my wisdom ends. I have 2 VPNs running via FreeS/WAN on the firewall host, and now I want to replace my hand made setup with shorewall. I use the same FreeS/WAN setup as it is working already. I read through the documentation but there are only examples of 1 static net2net VPN and 3

2 things, first an idea... next a question. QUALITY MEASUREMENT IDEA: I find it difficult to hear 2 voice samples and tell which is nearer the original, especially if the background hiss is slightly different. So what if you actually subtract the post-compression sound from the original and then listen to the DIFFERENCE. If you can't hear any voice except background noise and some hiss from

Hi, I have speex running as a part of a voice conferencing app. Well, one under development anyway. I'm running VBR at quality 3 and get a "hissy-squelchy" background noise. This is fine, kinda, because the internal microphone in the laptop picks up hiss, the sound of the (actually very quiet) hard drive and generally speaking is of less than exemplary quality. To help

I use 3 Ubuntu server 14.04 as KVM guest. Now I run 5 tinc vpns in switch mode. I use the same key for every vpn. Every kvm has connect to the two other servers. I need 5 vpns to have 5 networks connected but seperated. 1 vpn was running fine, but now that I run 5 vpns, my kvm crashes sometimes. Especially when I stop tinc. I used tinc 1.0.26 from repo. Now I tried tinc 1.1 pre, same problem but

I think you misunderstood my quality measurement idea. I mean if you subtract the original and the one after, the LESS voice that is less over or the LESS you can tell when someone is speaking, the better the compression. This is still subjective but I think its easier to tell this way because its easier to tell how much voice is remaining than to tell how much the compressed voice is missing from

I have clients, which connectin to Internet through vpn. I want to dynamically share bandwith between vpn connections, so if there few connections, then they get all bandwith, if more then they get their minimal guaranteed bandwith. my idea is: ip-up.local: tc class add dev $DEV parent 1:1 classid 1:2${1/ppp/} htb rate $[$RATEUP/$VPNS]kbit ceil ${RATEUP}kbps tc filter add dev $DEV protocol ip