similar to: ssh over udp (or: -L option listening for traffic with a UDP service?)

Mike, As it stands right now, we use xz for our compression, so if rsync had a similar option for xz that would probably be an improvement. However, I think being able to decompress directly to the remote system would save more; elsewise I don't see how I'm going to be able to avoid untarring and uncompressing the whole xz file on the other end. thanks much, ed On Tue, Dec 6, 2016 at

So I'm using a WRT54GP2-NA when I travel, as I travel alot, to give me a phone at my hotel rooms, etc. During the day or late at night the thing works great - best ATA I've ever used. However, in the mid-evening (when many business travellers are at the hotel room doing work), the outgoing audio channel gets so choppy that the person on the other end can't make me out clearly.

Hey all, I''ve got an interesting one for anyone who''s up for a challenge. Essentially, I have a very choppy effect, that almost looks like timeouts are overloaded or interfering or something, that only occurs when sortables are on the same page as "standard" effects. Here''s what I''m doing: I have a menu that slides in and out on the right side of

I am experiencing a situation where I am getting intermittent choppy audio. Here is the network layout: Termination provider -> IAX2 over the Internet -> 20Mb fiber connection -> router -> Asterisk My ATA connection goes into the router between the fiber and the Asterisk server on another interface here is the layout from me to Asterisk: Sipura ATA (SPA1001 running

Hello, I'm new to the list, just want to say thanks a head of time to who ever can help me out with some streaming problems Hareware/software: Athlon 700 128megs of RAM 256megs swap FreeBSD 4.2-RELEASE 100mbit connection icecast 1.3.10 ices (beta) My setup: I am running 2 spawns of icecast and 2 spawns of ices. one ices process is reencoding to 24bit and the other to 128bit and both are

hey.. I talked to my sysadmins about getting access to the dtrace_kernel role, and they said they were hesitant to give this out because they thought it was a security risk - ie: that you could use it for privilege escalation. How true is this? Is there a way to make it user safe? If not, why is it offered as an option for regular users? Thanks much, Ed -- This message posted from

I've setup asterisk on a Dell Poweredge 1300 Server. PIII 600, 500mb Ram, SCSI HD's. It is running Fedora Core 1. I have 3 X100P cards and 6 Grandstream IP phones. Everything works but the sound quality with the FXO cards is poor. Static and choppiness. It works great between the IP phones. The only way I can get good sound quality with the FXO cards is to start * while in a terminal

Hi all, It appears that something very strange is going on... Here's the deal -- whether someone calls in to my * server or I call out (doesn't matter), I can hear them perfectly: no gaps, no packet loss, nothing ... however, when I speak there seems to be very noticable latency and "choppiness" as if there were packetloss or lots of jitter. I'm using SIP for outgoing

Hi all, I've been testing various codecs to eliminate "choppiness" that I sometimes get on my Asterisk IAX2 <> DSL <> provider (Exgn) connections, and Speex seems to work the best, so far - but Speex seems oddly unpopular. Can anyone share their experiences with Speex (good and bad)? Is anyone using it in a production environment? I like the variable bit rate and

I'm getting some sort of weird distortion in the sound being output. I've tried many variations (including copying the seeming working code straight out of decoder_example.c) and still it comes out distorted. It is a different type of distortion from choppiness (which it also has). The easiest way to hear it is to pause play for about ten seconds and then resume. xmms will continue to

hey all, I'm using openssh-3.5p1, was trying to set up a 'reverse telnet' session (sun solaris 2.6 on both machines). Anyways, I was doing: server% ssh -R 1111:<server>:2222 <client> client% ssh -p 1111 <client> where <server> is behind a firewall and <client> cannot reach <server> Anyways the idea was to connect to the socket on

hey, I was wondering - is there an option to rsync which allows you to exclude transfers of files with a certain size or other characteristics? I want to setup a 'partial' rsync mirror where I only pull files under 5M, for example.. I see the option for including/excluding file names in the usage, but not file sizes.. Ed

hey, I just got an ugly, hard-to-fix surpirse when I upgraded from 3.6 to 3.7 for openssh - Now, in 3.7.1, /etc/default/login overrides --with-default-path! Before, it used to prefer --with-default-path (as configured). I'm hoping that this is a bug. It forces every user to have an appropriate .login,.cshrc, or whatever *or* to comment out the appropriate entries in / etc/default/login!

pstn -> sip gw -> * -> C7960 When I dial into * via the pstn, I hear the ivr menu just fine (good quality). I press 3000 (valid extn), and I begin to hear ringing however the ring back is very very choppy. I answer the C7960, and speech is clear in both directions. Place the C7960 extn on hold, and the MOH is very choppy. Checking 'sip show channels' indicates both the sip gw

hey all 'scp -r ' follows symlinks. IMO this is a bug and should be changed - it: a) hampers the use of scp. As it stands, I cannot use 'scp -r' because of this behavior. If someone links to '/', or if I hit a recursive symlink, I'm screwed. b) It is inconsistant with cp. When you 'cp -r' on a file, it does NOT follow the symlink. When you scp

Or just create bash (or whatever your favorite shell equivalent is) alias? alias issh=?sah -F ~/.ssh/altcinfig? Or wrapper scripts in ~/bin/ you can invoke? Literally what I do for mosh, mosh+screen etc. -- "Genius might be described as a supreme capacity for getting its possessors into trouble of all kinds." -- Samuel Butler On Fri, May 3, 2024 at 20:27 <openssh at tr.id.au>

I just signed up with livevoip for 800 DID and have very choppy audio. From PSTN to my asterisk is ok but asterisk to PSTN is terrible. I am using IAX and was assigned to server iax01.nyc.*. I do not believe it is a bandwidth problem on my end and I have no problems using iax with gafachi. I opened a ticket with livevoip but no response yet. Would I be better off using sip with them? Is there

Hi guys, I have the following configuration: 10 SIP softphones <--> Asterisk <--> PSTN Audio is always good on SIP softphone side, but callers from PSTN side *sometimes* complain that the audio quality is bad (and volume low). The QoS is turned on on the computers where SIP softphone is installed, and the tos setting in sip.conf is set to 0x18. The interesting thing is that usually

Hi guys, I am planning to connect two Asterisk boxes that are currently running in two different countries, using IAX. I was wondering if anyone could provide me with some links or suggestion regarding best practices in connecting two Asterisk in such way. I guess many of you have already tried this, and already have some know-how (what I should be careful about, what to avoid, etc...)?

Hi I am looking at purchasing some DID lines from Teliax to install it on my asterisk. i would like to know some feed back on "Teliax" before i purchase. suggest me if there are better sevice providers. thanks Giridhar Bandi -------------- next part -------------- An HTML attachment was scrubbed... URL: