similar to: Newbie question: Poor transmission quality

Mark Glassberg schrieb: > I am using Icecast2 with Ezstream, sending a playlist of mp3s. The signal > quality is fine over my home network; but out on the internet it is poor, > with frequent interruptions and losses of connection. > Sounds like network problems. > I would like to know which of the following two conditions, if any, can be > causing the problem. First, I

Hi All; I just need to know the openion about Grandstream phone, actually I tried Budge Tone 201 and I chocked that there is a noise in the handset (zzzzzzzzzzzzzzzzzzzzzzzzzzz) always, but in the speaker the sound is good and no noise. Anyone has idea about Grandstream, and if they have a lot of problems and such noise in handset? Or my luck was bad that this phone is defected? Regards Bilal

Hi All; I have been chocked just when I saw some posts talking about how much the IAXy is bad :) - So I would like to ask, did any one try it later and wether it is good or not? I am asking this because I need to use it as it is NAT Transparent (as I read also, and I did not try it to see how much it is transparent). What about codec? Why it is only support g711 and does not support compressed

> 2016-08-18 12:39 GMT-04:00 correomm <correomm at gmail.com>: > >> This bug is reported only on the VM's with CentOS 7 running on on VMware >> ESXi 5.1. >> The vSphere performance graph shows high CPU consume and disk activity only >> on VM's with CentOS 7. Sometimes I can not connect remotely with ssh >> (timeout error). >> > I'm

On Thu, Aug 9, 2018 at 4:26 PM jan Vitek <vitekj at icloud.com> wrote: > > > I'm now confident that I > > can avoid using "type" by itself, and instead always use it in a > > compound phrase (like type system) to avoid confusion. That leaves the > > `.type` argument to many vctrs functions. I'm considering change it to > > .prototype,

Hello, I have the go ahead to fix documentation. For starters I will clean all extraneous crap from the markup. Then fix any errors and typos. If anyone sees any gramatical errors or typos or needed clarifications or expansions or whatever, please post them here. Or additional FAQ entries, etc, all will be considered. I hope to keep most of this conversation on the shorewall-devel list. Any

> > As Gabe mentioned (and you've explained about) the term "type" > > is really confusing here. As you know, the R internals are all > > about SEXPs, TYPEOF(), etc, and that's what the R level > > typeof(.) also returns. As you want to use something slightly > > different, it should be different naming, ideally something not > > existing yet

Hi, I'm running CentOS 6.5 as NFS server (v3 and v4) and exporting Ext4 and XFS filesystem. After many months that all works fine today the server crash: Jan 30 09:46:13 qb-storage kernel: ------------[ cut here ]------------ Jan 30 09:46:13 qb-storage kernel: WARNING: at lib/list_debug.c:26 __list_add+0x6d/0xa0() (Not tainted) Jan 30 09:46:13 qb-storage kernel: Hardware name: PowerEdge

Hi, does anybody set up an icecast server delivering streams > 160KBit/s ? I tried that with liveice and shout but in both cases the stream stutters very much. So, I have eliminated the Net (stream to localhost), the CPU (P3/800 should be fast enough), and lame (can encode 320kBit/s 3 times faster then the wav has secs). So the Problem should be icecast. can anybody please enlight me?

Hi,all When we allocate bandwidth to a host/device in LAN , how can we know the max. bandwidth the device(e.g.1394,ethernet,powerline..) can receive(in the idle condition, no other application occupy the bandwidth) ? Can RSVP signaling do that?or others ? Thanks Fionna ___________________________________________________ 最新版 Yahoo!奇摩即時通訊 7.0，免費網路電話任你打！ http://messenger.yahoo.com.tw/

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Hi, While looking for the cause of disturbance in call I found this error coming in console RTCP SR transmission error, rtcp halted Google search only shows some bug reports relating to MOH and Hold. What could cause this message? Could this be a symptom causing call disturbance? Where should I start digging to find out the reason for this error? I am using Asterisk 1.4.19 with zaptel 1.4.9.2

I have a situation here where a user has an AAstra 480i phone, which function corectly. The phone is behing a nat-router (a linksys wrv200 for it's VPN point to point facility). The phone is plugued in a port wich has qos enabled. And when the user places a call, sometimes (not always), we get this in the console : [May 8 13:41:55] WARNING[5804]: chan_sip.c:1948 retrans_pkt: Maximum

Hello listamtes, Here's is a curious issue: netcat (nc) seems to not do UDP for me, even though TCP works flawlessly. Not even on the local host. Here's the "server session" I am running: cat test.dd | nc -l -u -n -k -v 2456 (test.dd is just a large file with random data) Here's the client: nc -v -u 127.0.0.1 2456 | dd of=/dev/null ^C0+0 records in 0+0 records out 0

Hi Folks, I used to run transmission for torrents on opensolaris snv_134. My working zfs was newsan/tmp and when the torrent completed i had transmission configured to move the files to newsan/incoming. I upgraded to sol11exp and the related zfs updates and now the files no longer move to newsan/incoming upon completion. I believe this to be a zfs acl issue and could use some assistance in

Dear friends - a couple of papers in PNAS (lastly:framework for making better predictions by directly estimating variables' predictivity, Lo et al PNAS 2016; 113:14277-14282) have focused interest on mapping complex traits to multiple loci spread all over the genome. I have been around on the relevant taskview(s) I hope but fail to see that the backward haplotype transmission association

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I'm having an intermittent problem with fax transmissions, something I call the streaking mascara syndrome. TxFax will be part way through a transmission when something fails--on the fax printer, part of the page will have correctly printed, but the rest of the page is "streaked" to the bottom. I have several successful multi-page transmissions interspersed with streaked

Hello everybody, I'm experiencing a strange problem with my Asterisk. I hope you can help: Asterisk is running at my company behind NAT. Ports 5060 and 10000-20000 are being forwarded to it. I have put the router's external IP-address into externip in sip.conf. At home I'm using an AVM FritzBox Fon WLAN 7050 which is registered with the Asterisk at my company. When I try to call

Asterisk doesn't seem to be relaying 183, Session Progress SIP messages received from an upstream host back to the phone. Anyone know why? Here's the SIP message that Asterisk receives, and it does nothing with it. It doesn't pass it back to the phone. <-- SIP read from xxx.yyy.142.234:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP