similar to: Problems with queue announcements under high call volumes

Hi, I''m looking at a system hang on a large box: 160 cpus, 2TB. Dom0 is booted with 160 vcpus (don''t ask me why :)), and an HVM guest is started with over 1.5T RAM and 128 vcpus. The system hangs without much activity after couple hours. Xen 4.0.2 and 2.6.32 based 64bit dom0. During hang I discovered: Most of dom0 vcpus are in double_lock_balance spinning on one of the locks:

? Do you have any references for me to investigate, I am trying to understand how noise is reduced by introducing higher sampling rates. (I tried to search, but maybe it is so obvious that nobody even explains it) This is not very obvious. It requires you to understand basic signal processing theory. I will give some pointers below. Any physical signal (e.g. audio coming out of speaker, current

Is anyone out there using SIP Witch in conjunction with Asterisk? It claims to be able to "enhance existing IP-PBX solutions such as Asterisk", so maybe it can be used as a simple means to provide secure/encrypted calls. GNU SIP Witch - Summary <http://savannah.gnu.org/projects/sipwitch> GNU SIP Witch - GNU Telephony <http://www.gnutelephony.org/index.php/GNU_SIP_Witch>

Hello, I saw some conversation about this in the archives, but nothing definitive. If a call comes in over a CO line via the TDM400P, the Comedian Mail recording volume is so low it's inaudible. Calls coming in via SIP or IAX do not have this problem. Does anyone have any information on this issue? Thanks, Adam The contents of this email message and any attachments are confidential and

This may expose my ignorance, but here goes :) I've been asked to figure out how much bandwidth would be needed to handle 1,000,000 minutes a month. Here's the environment: ) All calls are received via SIP. ) All calls use the ulaw codec. ) Calls average 10 minutes in duration. ) The "busiest" hour will account for 10% of the daily total. This is how I'm figuring

I'm rebroadcasting a realplayer stream and there are two problems. The icecast2 sounds tinny and its delayed something like 3secs. Any advice to perhaps use speex as it's talk radio? Can I cut the delay? Here is my ices2 config: http://static.natalian.org/2006-06-21/ices-alsa.xml Here is details of the feed, so you can take a listen:

G'day ... Altho it is rare, we are getting union_lookup panics on our server(s) ... as most on this list already know, we make heavy use of unionfs to share between jails, and I already know that this is the "kernel side" cause of the problem ... but ... From what I can tell, though, the "trigger" for the panic is pkg_delete ... if I build a jail'd environment,

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 trying to encode audio tracks as ogg vorbis at very low bandwidths for peer to peer broadcasting. I.e. need to have two streams uploading from modem users, which means the bitrate needs to be around 18-22 kbps. It's not easy to get good quality at this bitrate. The recipe I'm using at the moment is: oggenc Mon-Sep-20-04-0\:3\:27.wav -b 20 -M 22 --resample 17000 --downmix -o

Hi all and thanks for reading. I am experiencing a frustrating issue with asterisk where on some calls the volume suddenly drops to inaudible o completely fades away for a time. If you hold on long enough (20 to 30 seconds) the sound will come back. My asterisk server is on a public IP, and basically acts as a VoIP bridge receiving calls from my customers (all of whom use Grandstream GXW400X

I am trying to use the speex echo canceler and preprocessor with sipXtapi to develop a sip user agent. the sipXmedialib call flow graph uses 10ms frames and I am not sure what the implications are if I try to change this. The documentation seemed to indicate that a 20ms frame was recomended, but it didn't go into the consquences of using other frame sizes. We are going to use headsets

Hi Guys, My first post here so hello to all :-)) Anyway, my question: Does anybody know if there is a tool out there that can process the icecast log files and return a graph of listener numbers over a set period? I'd like to be able to know when our most busiest time was. Thanks very much in advance, Andy

Hi, All, I have a question about agents and queues. Right now we have about 4 queues in our system. Some agents are in multiple queues. Our main queue is for technical support and it's by far our busiest queue as well. We have the autologoff feature set to 14 sec right now in the agents.conf file. The problem I'm running into is we don't want people in our sales queue (who are

Hello Everyone, I can see that a few people are interested in SIP WIFI phones. I have tested several Linksys 300,and it is OK. More of a toy then a business tool. It a poor built in ear speaker, which makes all calls sound tinny, and the unit is known to hang. I have two Linksys 300's that are fun to play with however, I wont hand them out to users. HOWEVER- The Zultys WIP 2 is an

List users, Using Asterisk in an inbound call center environment has led us to pushing the limits of vertical scaling. In order to treat each caller fairly and to utilize our agents as efficiently as possible, it is desirable to configure each client as a single queue. As far as I know, Asterisk's queues cannot be distributed across servers, so the size of the largest queue we service

The results are at www.rational.co.za/speex.csv Each of the 11 quality settings is tested 3 times (narrow, wide and ultra wide band). Strangely narrow band quality 11 outperforms all wide band tests, but it can be due to my subsampling or some other inaudible effect like delaying. You can import it into Excel and sort it by SNR or other value. Divide the bits by 24 to get the bps. The

I'm attempting to use speex preprocess for automatic gain control in an application I'm working on and could use some help. I'm using Opus as my codec. In order to keep the number of packets down, I'm using 60msec frames. I'm sampling at 48KHz as is recommended for Opus. So, the frame length is 2880 samples and the sampling rate is 48000. The source of the data is a

Hello, I'm not a programmer or Opus developer but I tried to test the sound quality of a music file (Nick Warren - Devil's Elbow, freely available from the author's Soundcloud account) encoded with libopus 1.2 RC1. I used Windows binaries from free-codecs.com. I noticed that in the case of my selected music file (which is generally harsh on lossy codecs as it's necessary to

All, We are in the process of trying to setup our network to use Verizon's SIP "trunking" product. They say that since Asterisk is not on their certified list of approved devices, we need to go through a field trial to get it approved before allowing us to use their service. Where we are at is getting the design approved. We are trying to watch our budget at the same time. We

Hi Opus list, Please forgive me if this has been asked before. I find that Opus encoder created in mode OPUS_APPLICATION_AUDIO (as opposed to _VOIP) is using a lot of bits to encode silent periods of speech. This is relevant to a voip application for which good quality music is desirable, and in which I add a minimal comfort noise (order of few bits loud, e.g. MLS signal of amplitude 1 or 2)