similar to: SIP dropouts

I have time-series data from a pair of inexpensive self-logging 3-axis accelerometers (http://www.gcdataconcepts.com/xlr8r-1.html). Since I'm not sure of the vibration/shock spectrum I'm measuring, for my initial sensor characterization run the units were mounted together with the sample rate set to the maximum of 640 samples/sec. Unfortunately, at this sample rate there are significant

Hi all, We have recently implemented an Asterisk system using Trixbox (asterisk v1.4.4 at the moment, yet to move to 1.4.9) but are getting pressure to switch back to our old key system unless we fix two major issues. So please help me avoid switching back! An overview: We have about 12 Linksys SPA941 SIP phones connected on a private switched network to our asterisk box which is a

First of all, *thanks* to all the developers for their hard work on Ogg Vorbis. It's greatly appreciated. I'm having a problem with encoding in Debian GNU/Linux 3.0. The oggs contain occasional dropouts and "blips" here and there. Example files: Original (~7 MB): http://personal.inet.fi/musiikki/nebularia/test/t2title.flac Ogg Vorbis file (~800 KB):

We are having some problems with the audio during a call dropping out for a second or so and then returning. This is not at the beginning of the call and seems to happen randomly. We use primarily Cisco 7960's and our * is a very current version of CVS head. The server itself is a Compaq dual PIII 800 w 1GB of RAM, it has a single T100P card with a T1 to our voice provider. We have very light

Hello List, I'm trying to prepare some lecture notes on non parametric methods, and I can't manually reproduce the results of the wilcox.test function for ordinal data. The data I'm using are from David Howell's website, available here http://www.uvm.edu/~dhowell/StatPages/More_Stuff/OrdinalChisq/OrdinalChiSq.html If I run the wilcox.test function on the data I get a p-value of

Hello, I'm having a problem with significant audio dropouts occurring in voicemail messages left via an ISDN-BRI trunk. Dropout durations are as short as 15ms and as long as 200-300ms. The audio that is recorded, appears to be otherwise complete, just with frequent holes punched in it. The same trunk has no problems with audio files played toward it from voicemail, nor interacting with

All, The users in our organization are well, quite frankly, sick of phone service that is being provided. The choppy phone calls, and drop outs are detrimental to our sales force. I've tried about everything I can think of. Moved the asterisk server from VM machine to dedicated machine More than enough bandwidth Setting 802.1p = 7 Set Dedicated voice traffic 35% of bandwidth. Not sure

hi all I keep getting 'bzzt'-sounding dropouts, but I've only heard them when from behind NAT. I don't think it's related to bandwidth, as I'm on a 2400/640 ADSL line. I'm currently using ALAW. Any ideas how to debug this? SIP DEBUG doesn't show me anything when dropouts happen, although I'd doubted it'd do so... How can I debug the RTP part of SIP?

Jay, I have had a lot of trouble with the FXO ports on Adtran TA750. Unless the incoming POTS lines have a balance impedance, they will buzz very bad. Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jay Milk Sent: Sunday, April

Hi there, I am using bri-stuff.0.1.0-RC2k and it seems that things didn't become better. I have got lots of dropouts on the IAX2 link (no matter if jitter buffers are enabled). Further the MP3Player application does not playback streams like http://somestreamserver/somestream. It stops saying: -- Executing MP3Player("SIP/27870-ba4f",

Is it possible to run a packet capture to see what happens over the wire when this occurs? That would give some concrete information. If you have the disk space, just rotate out a capture file every hour. Once you catch it, filter out a pcap file with what happens when the sources are disconnected, and send us a link. Brad Isbell brad at musatcha.com http://www.musatcha.com On Tue, Nov 4, 2014

hi I keep getting sound dropouts during SIP re-registration, and I can't find a remedy for it. I use SIP friends from MySQL. Below is SIP debug output for the re-registration Thanks in advance roy ------- *CLI> sip debug ip 80.202.161.221 SIP Debugging Enabled for IP: 80.202.161.221 *CLI> Sip read: REGISTER sip:sipgw1.briiz.no SIP/2.0 From:

I've been having some problems getting a channelized T1 working with a T100P card. Perhaps someone can help: I have an Eschelon T1 coming into a Vina Integrator box. This box splits out the T1 into an ethernet plug for bandwidth and a secondary T1 which I plug into the T100P card. I've connected the two with a T1 crossover cable. I get a green light on the t100p card and can make

It seems that each time I get a new checkout of * from CVS my Cisco 7960 works worse than before. I know this stuff's in flux, so I mention this in case it's news. Anyone else having trouble? What I'm seeing (er, hearing) is really choppy audio. The previous version I had installed had fairly frequent audio dropouts (not present when I make the same calls through the same * box

Hello All, My asterisk installation has now been running for over two months without a hitch, and I've decided it's time to move things around a bit. It's currently installed on a 2.7GHz Celeron under RH9 installed on a 10GB "leftover" drive. Thanks to the strange marketing method called "Mail-In-Rebate", I have a fresh 160GB drive ($50), and I'm itching to

Hi all, I am a biostatistician and I have developed my own ranking system for clinical data. I would like to test the efficiency of it w.r.t. to other ranking systems. I would like to simulate the data and after assigning ranks to my observed scores(after neglecting dropouts), observe the type I error. If I want to do a Kruuskal Wallis type of test, what test statistic should I use to test for a

Does anyone have the solution for this. I have a T100P going to a channel bank for all internal extensions. Calls from one internal to another sounds prefect. When placing a outbound call (going out over X100P) there is a very slight echo, almost like the comfort noise is delated by a fraction of a second. Its just a little annoyong, but when moving papers on my desk, it is very

I need to debug a call quality issue with remote users on the other end of a satellite link. The symptoms are: we here on the Internet side can hear them just fine. On their end, things work sorta OK most times, but they often suffer from severe dropouts and digital warbling, both of which I attribute to them missing packets. Often times they can't make out a word we are saying while we can

Hey all. This morning I found that my audio playback is randomly sprinkled with sound skips and dropouts. I went to /var/log/yum.log and found this: Feb 09 20:18:22 Updated: lame-3.99.4-2.el6.rf.i686 I'm not saying that caused the problem but it's all I could find that changed. When I ls in /dev there is no dsp entry. That would explain why mplayer Can't open audio device

After downloading the latest CVS head and testing it with the Cisco 7960 (SIP v7.2), it appears that it's still necessary to patch rtp.c to avoid audio dropouts. I'm quite sure my gateway provider is running an older version of Asterisk, and I suppose that this may be the root cause. But I mention the issue here because it seems like it would be a mistake to ship Asterisk 1.0 if it