similar to: Dropouts and echo

Hi List, Does anyone have experiences to relate on the various Asterisk-based PBX appliances out there? Like the Aastra 160, Digium S844i, etc. Do the Epygi Quadro and Grandstream GXE also use Asterisk? Thanks, Femi -------------- next part -------------- An HTML attachment was scrubbed... URL:

Howdy all... When making SIP calls through my X100P from X-Lite to the PSTN I'm getting 3-5 second dropouts in both directions. I've tried ulaw and GSM, but that doesn't seem to make a difference, and the * box is on my local net. Here's my hardware: Celeron 2.4GHz, 512MB, Slackware 9.1, 2 X100P, 1 T100P. Any ideas what could be happening, or pointers as to how to shoot this

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):

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

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 list, Trying to upgrade someone's workstation here to 5.2 (was installed from a 5.0 DVD I think). The RPMs on our internal mirror are in-tact and pass a 'rpm -- checksig' test, yet when I run a 'yum upgrade' a large portion of them are corrupted and fail the GPG check. This seems to be isolated to yum, as downloading the RPM directly via FTP with wget or lftpget

Hi all, New to the list and (mostly) new to CentOS. We have just moved from running RHEL to CentOS on our main production servers. Anyway, I am configuring a couple of new SunFire v20z servers here at the moment and trying to get the systems' management interfaces to work fully under CentOS. First issue was trying to install the openipmi RPM from the CD that came with the SunFires, this RPM

Does anyone on-list have any experience with this device? It's a 6 port FXO<>SIP gateway. Michael Graves -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc. mgraves@mstvp.com o713-861-4005 o800-905-6412 c713-201-1262 fwd 54245

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

Hello everyone and Timo in particular, about a year ago I implemented a SHA-1 variant of the HMAC(-MD5) present in Dovecot. I had always disliked this a bit, because it replicates a lot of code. This patch generalizes the HMAC function to take a hash_method struct as parameter, and changes existing code which uses the "old" HMAC function to use this new one. I'm not really 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?

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",

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

> In my case, I have good incoming quality and terrible quality going out. > That is, I can hear people perfectly well but they complain that my > voice drops out and is garbled regardless of who places the call. This suggests to me that you may have congestion problems in your "upstream" traffic flow. Setting QoS on the packets may not help, if whatever router you are using

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

Hello, ? I've searched and searched for a solution to this problem but continue to come up empty. ? I have a very generic setup.? CentOS Linux 7.3.1611 running on AWS.? ? I installed the QMail toaster from here: http://www.qmailtoaster.com/ ?( I had been using Bill Shupp's Qmail Toaster for years and years, but he stopped supporting it.) ? As such, I did have to convert my

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

Hi list, The bug reporting guide in the wiki mentions we should notify the - devel list when reporting a new bug, so I just wanted to bring some attention to: http://bugzilla.xensource.com/bugzilla/show_bug.cgi?id=1328 I''d reported another bug myself (http://bugzilla.xensource.com/bugzilla/show_bug.cgi?id=1332 ) and marked it as a duplicate of the one above as they seem to be the

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:

Hi list, Does anyone have any advice on the following: Incoming calls to our office come in on a SIP trunk. Since all our offices/desks are in close proximity, we would like just a single phone to ring when a call comes in instead of ringing every person's phone. Currently we've got this working by having all the phones in a callgroup/pickupgroup and incoming calls ring the