similar to: Strange dropouts

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, ince I'm currently working on a program that manages all my music files and am working on Ogg Vorbis support right now, I have two questions: 1. Should Tag names (i.e. "ARTIST=") be lower- or uppercase. I've encountered both and right now I automatically uppercase everything I get and then write it to the file that way. 2.

Simply put: When will Oggenc accept input files with higher resolution than 16-bit? Thumbs up to all the developers - the sound quality of Vorbis just keeps amazing me! I only use Ogg Vorbis when distributing my music today. Cheers, - Tomi <p><p><p><p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To

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

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

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

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 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

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 This is a feature request. 1) Make sshd to ignore garbage that may appear before ssh identification string is received. Such "garbage" may be for example telnet negotiation codes. This should be pretty easy task. 2) Make ssh to work in cleartext mode (and have minimum telnet negotiation handling) before it receives ssh identification string. This requires somewhat

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OK, I have been banging my head into the wall on this now for two or more weeks We have a samba server (2.2.8 then 2.2.8a) running on top of a 320 GB raid ATA-raid array with a FastTrack controller. Both the server and workstations are running through Gigabit NIC's. Every five or so minutes the workstations loose communications with the server, Then it takes 10-15 seconds for the

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 looked at your network diagram. Try checking the configuration of the > Ethernet ports on the firewall and the Asterisk box. Make sure they are > set to auto-negotiate and not set to a fixed speed and fixed duplex. > I have found in the past that if one end of a link is expecting auto- > negotiation (as the switches probably are) and the other end is expecting > a fixed

Hi, We have a Ericsson BP250 Phone system setup witht he following configuration Telco <-> Asterisk E405P <-> BP250 The system seem to work perfectly on 1.0.9 for a very long time but there is some functionality we wanted to take advantage of in the 1.2 version branch so we upgraded. Currently running Asterisk 1.2.4 Zaptel 1.2.3 (noticed 1.2.4 is out will upgrade next

The following is a message to be sent to the President of the United States of America. Although we may not be able to do a great deal from where we are, but for the people of America just knowing we care and feel their sadness will help. Please put your name on the following list and send it to all you know and who care. If you are the 100th name and every 100th there on could you please also