similar to: Tired of dropouts and garbled phone, calls - where to go next?

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

> > Different brand/model, but similar as they are both el cheapo, > entry-level headsets. I tried using them on a laptop, and I get > marginally better microphone output, even with its volume cranked all > the way up + automatic gain control enabled. > > I guess those on-board soundcards by Realtek aren't as good as a > quality microphones. I'll get a USB headset

On 15/12/17 07:05, Kenneth Porter wrote: > I came across this on the Fedora devel list. I added > /etc/sysctl.d/51-bufferbloat.conf containing the suggested line and it > installs the new codel qdisc as desired. There's probably more knobs > that might be useful to tweak but this makes a good start. More reading > on the bufferbloat site suggests that the later "cake"

For some reason a couple weeks ago users began experiencing garbled audio in one direction when dialing out via our VoIP provider. This happened at multiple sites simultaneously. The VoIP provider doesn't think it's their problem. If I switch to another codec so that Asterisk transcodes everything is fine. On conference calls (where Asterisk gets in the middle to relay ulaw to all

problem - occasional garbled calls, mostly remote users. T1 connected to PSTN, SIP over local LAN and internet to "remote users". NAT at local firewall and at remotes. There is no traffic shaping in place, no QoS. Most are Polycom phones, two Aastra's. Start with QoS on LAN switches? No 2x4's please, start with 1x4's. joe a.

> 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

OK, I have windoze ME installed on my system and have been trying to run IE 6.0 with wine release 20010824. Trying to start iexplore.exe goes along until: Unhandled exception: page fault on read access to 0xdeadbeef in 32-bit code (0xdeadbeef). In 32-bit mode. 0xdeadbeef (_end+0x9df10793): *** Invalid address 0xdeadbeef (_end+0x9df10793) -- no code -- Enter path to file 'lstat64.c':

Hi all, What is Bufferbloat? http://gettys.wordpress.com/bufferbloat-faq/ Maybe this kind of discussion will bring out the John Todds of this world, I can only hope and dream: Bufferbloat: http://www.voipusersconference.org/2011/bufferbloat/ Call in and talk to Jim Gettys, who co-developed X Window System and was a part of HTTP/1.1 - this is someone we'll all be proud to meet, and you can

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

I came across this on the Fedora devel list. I added /etc/sysctl.d/51-bufferbloat.conf containing the suggested line and it installs the new codel qdisc as desired. There's probably more knobs that might be useful to tweak but this makes a good start. More reading on the bufferbloat site suggests that the later "cake" module will be even better, but it requires a newer kernel

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",

Can you explain? Sent from Samsung Mobile <div>-------- Original message --------</div><div>From: Tiago Geada <tiago.geada at gmail.com> </div><div>Date:03/07/2014 9:04 PM (GMT+02:00) </div><div>To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> </div><div>Subject: Re:

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):

I'm deploying a CentOS 7 box as a gateway and I'm trying to figure out how to set up traffic shaping. Historically I've used the Wondershaper script but apparently it's not deprecated in favor of superior queue management. I haven't yet found a packaged solution and I'm wondering what others do to configure this kind of thing. Apparently the new modules are available

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

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: