similar to: moving WiFi phone

Hello, I've found GlusterFS to be an interesting project. Not so much experience of it (although from similar usecases with DRBD+NFS setups) so I setup some testcase to try out failover and recovery. For this I have a setup with two glusterfs servers (each is a VM) and one client (also a VM). I'm using GlusterFS 3.4 btw. The servers manages a gluster volume created as: gluster volume

Good morning, On Mon, 2019-05-06 at 08:44 +0000, Jürgen Bund wrote: > Hello again, > thanks for your response. > > I have still some questions: > I want to stream a 24/7 radiochannel with mp3s. So I can't send all mp3s with one http PUT request. > So is there a possibility for that with http PUT? > If yes, how can I send the next mp3, so that icecast is playing it

We're looking at a large wifi phone deployment, and we're looking for wifi phones that: 1. Are SIP compliant (Asterisk friendly) 2. Provision capable (ideally TFTP of own MAC address) 3. Industrial quality (no cheap plastic stuff). 4. Well documented (and none of the "only telco's get documentation" crap) Does anyone have a suggestion? Thanks, MD -------------- next

Hi - I have a ftp server running version 2.0.7 of vsftpd on a CentSO 5.2 server using iptables behind a Linksys router. The setup works for UNIX machines on either side of the Linksys router. For the Windows machines it only works if they're behind the Linksys router - ftp does NOT work if they're outside the Linksys router. I'd like to solve two problems: (1) make ftp work

I'm gonna CC the swfdec mailing list, as I think it's interesting for everybody thinking about Swfdec on embedded. > I have cross-compiled swfdec0.7 on directfb, and the swfdec-directfb-player > can work on NXP platform , > > The CPU cost is high, video can be viewed continuesly , however no audio > output. > The DirectFB player was the result of a project we did to

Hello again, thanks for your response. I have still some questions: I want to stream a 24/7 radiochannel with mp3s. So I can't send all mp3s with one http PUT request. So is there a possibility for that with http PUT? If yes, how can I send the next mp3, so that icecast is playing it fluently (continuesly)? If not, is there a documentation for a TCP Socket Communication? Libshout is a c++

Good afternoon, On Fri, 2019-05-03 at 12:24 -0400, Fred Gleason wrote: > On Fri, 2019-05-03 at 13:19 +0000, Jürgen Bund wrote: > > I’m writting a source client in c#, in which I’m sending chunks of a > > mp3 file with http Put to IceCast. > > My problem is, that it is not a continues stream. Each http PUT > > request is an extra track. > > How can I generate an

Hello again, for now I figuered out to send a mp3 stream to IceCast with TCP Socket communication. Now I'm challenged with the correct timing for the data packets. Are there any API calls for getting the state of IceCast's buffer? I'm programming now on a scheduler for sending the mp3 frames, depending on the metada of each frame. I hope, that this is precisly enough. Can you tell

Hi all, find severel topics but no solution. Play Warcraft 3 on the network. ~15 min after the Dota match start Warcraft 3 crashs with following error. Code: root at tbitch:/media/Volume/Warcraft III# wine Frozen\ Throne.exe -opengl err:ole:CoCreateInstance apartment not initialised fixme:advapi:SetSecurityInfo stub fixme:d3d:IWineD3DImpl_FillGLCaps OpenGL implementation supports 16 vertex

Hi All We have the following problem: We configured samba with LDAP and this works fine. As soon as they try to login wit a user who is member of 15 groups, it takes very long to login with Windows and then an mostly an errormessage appears. On win 2000 is the error: There has been made a change to the server. Contact you sysadmin When a user logins (member of 15 groups) ldap shows the

Hello. I have a strange problem. Its not possible to pickup a call that was placed with a Siemens SL75 Wlan. When this phone calls an internal number and i try to pickup (*8) the call from my phone i get nothing. It seems i have the call for one second or so but after that the call is being cancelled. No problems with other phones (polycom, grandstream). Attached the complete sip debug log

Hi List, Hope everyone is recovering from the festive season :) (ok we still have new years i guess!) Anyways, I was wondering if anyone has had any successful dealings with WiFi phones and operation with '*' at all? I've been keeping my eye on the LinkSys WIP330 ( http://preview.tinyurl.com/nccxn ) and wondered your collective thoughts? Would I be correct in thinking that (as

Hi, I've been working on migrating my asterisk from zap to sip (due to compatibility issues between my TDM400P and my Hauppauge PVR500). I've purchased a Linksys SPA-3102 and a Siemens Gigaset SL75 WLAN (wireless SIP phone). I managed to get it all working with my asterisk 1.4.0 installation, but I'm seeing some interesting things with the canreinvite option that I can't explain,

Hi *, Does anyone have a lead on a Nagios plugin that speaks IAX or a small app to do so? I'm trying to set up remote monitoring for my Asterisk server and only IAX2 traffic is allowed through the firewall. Simply using check_udp to port 4569 yields no usable answer and Asterisk complains about receiving a midget packet or something like that :) jens

How do I solve the problem with between patterns: _1800 _1NXX I would like all numbers 1800, 1877 etc to go through iaxtel but all other numbers 1xxx via voipjet Example in my extension.conf I have: [iaxtel] exten => _1700NXXXXXX,1,Dial(IAX2/xxxx:xxxxx@iaxtel.com/${EXTEN}@iaxtel) exten => _1888NXXXXXX,1,Dial(IAX2/xxxx:xxxxx@iaxtel.com/${EXTEN}@iaxtel) exten =>

I'm hoping someone can help me with a problem I've been having for a while now. I've googled and wiki'd to no avail. Whenever I place an outbound call from * to a PSTN through a SIP or IAX provider (e.g. Voicepulse or Broadvoice), the first 1/2 to 2 seconds of the remote call are clipped (muted). For example, if I call a remote voicemail system that usually answers with

I am just now investigating Asterisk. Can Asterisk provide 6-10 party teleconferencing when configured properly?

Hi everyone, I'm trying to buy a small form-factor PC system for use with Asterisk and Hylafax in conjunction with a Eicon DIVA Server single-port ISDN card (needs full-size 5V PCI 2.2 slot, but PCI-X compatible). Use is very light - at most a single call at any one time. If the Mac Mini had a PCI slot I'd try to use that one, but oh well ;) Would anyone have some brand names or

Hi, I install CenOS 3.4, how import the gpg key ? Thx

I've been working my way through the softphones listed on voip-info over the last few weeks and I've not really found anything to fit the bill. Has anyone had more luck? The environment is a small call centre of 5 users. Operators often need to be able to transfer calls to other operators with different specialties, so the softphone needs to be easy to use and quick to transfer calls.