Displaying 20 results from an estimated 100 matches similar to: "FXS vs. FXO"
2002 May 13
1
Synchronizing uids on deferent machines running winbind
Hello,
I've a small Linux network, NFS amd NIS server with three clients. Each machine is running winbind to connect to a windows 2000 domain. All is working fine, I just have a problem synchronizing the uids generated by the winbind to a windows user that logs on to the Linux network. For example, on one machine a user can get uid 10000 and on another machine the same user can have uid
2002 May 13
1
Synchronizing uids on deferent machines running winbind
Hello,
I've a small Linux network, NFS amd NIS server with three clients. Each machine is running winbind to connect to a windows 2000 domain. All is working fine, I just have a problem synchronizing the uids generated by the winbind to a windows user that logs on to the Linux network. For example, on one machine a user can get uid 10000 and on another machine the same user can have uid
2003 Jan 15
3
winbind LOGON_FAILURE
Environment:
Redhat 7.3
Kernel 2.4.18-4
Samba 2.2.7
NT4 Domain
I have replaced the actual domain name and the name of the PDC with words
DOMAIN_NAME and PDC_NAME. In both cases, the correct domain name and pdc
name are used.
I removed, waited, and then re-added my linux server to our domain from the
server manager on our PDC.
When I then run smbpasswd -j DOMAIN_NAME -r
2007 Jul 19
8
Blank Voicemails
Hi, we're running Asterisk 1.2.10 and have been randomly being left
blank voicemails with long messages that we can't hear.
I've searched and searched but cannot find a solution.
This is what happens:
Internal Server runs Asterisk 1.2.10 where our mailboxes are
Incoming Server (behind a firewall) runs Asterisk 1.2.13 and calls are
bridged between this server and our internal server.
2013 Nov 06
3
Calling 'virsh' from within a hook script - avoid deadlock?
I'm trying to create a hook that attaches several virtio-scsi disks to a starting vm,
and it seems like the hook enters a deadlock while attempting to do so.
Is there any workaround around this? Any better way, forking virsh execution somehow?
Could anyone explain why this is creating a deadlock in the first place? Isn't libvirt multithreaded?
Thanks for the help!
Shlomi
### hook
2013 Nov 07
2
Re: Calling 'virsh' from within a hook script - avoid deadlock?
On 06.11.2013 16:17, Michal Privoznik wrote:
> On 06.11.2013 11:23, TSADOK, Shlomi (Shlomi) wrote:
>> [...]
>
> http://libvirt.org/hooks.html#recursive
>
> Do not do this.
>
> Michal
>
Maybe I should explain a bit why this is not supported. It's true that
libvirt is multithreaded. However, not in terms of multiple write/modify
accesses to a single domain.
2005 Oct 02
1
Adit 600 FXO card sound quality
I have an adit 600 with one fxo card connected to a Digium single span T1 card.
CallerID, disconnect supervision work perfect, however the users
complain that they have some sound quality issues, after testing it I
realized that whenever one is in a phone call they get like silence
between the sounds coming from the other party, almost like a cell
phone, in other words if there is no sound coming
2008 Jan 11
10
[Bug 14025] New: The X11 / X Server Exhibits Jitters in the Display after it is on for many hours
http://bugs.freedesktop.org/show_bug.cgi?id=14025
Summary: The X11 / X Server Exhibits Jitters in the Display after
it is on for many hours
Product: xorg
Version: 7.2
Platform: x86 (IA32)
OS/Version: Linux (All)
Status: NEW
Severity: normal
Priority: medium
Component: Driver/nouveau
2006 Nov 27
2
Shares problem
I am trying to produce 2 kinds of shares:
[homes] the typical home directory share.
And an all users (even guest) writable share.
On the home directory share, I am able to create files through my windows
XP, but I unable to create new directories.
The all available share I can't even browse.
Help please.
Shlomi
PS: I attached my smb.conf
2003 Nov 05
6
recording calls
Hello,
You can use ZapBarge as an extension in your dialplan to listen in on
conversations going on in Zap channels(Zaptel device channels)
As for recording you can use the Manager interface command StartMonitor to
start recording of a Zap channel and StopMonitor to stop it.
Zap channels are pretty much the only ones right now that you can directly
monitor and record through Asterisk.
If
2004 Sep 13
2
unavail and busy.
Hi guys,
What is different and the "context" to play unavail message and busy
message?
When a SIP connection is unregistered, voicemail will play unavail message,
right?
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2005 Feb 10
1
One "200Mbps" virtual link between 2 ethernet adaptators of 2 linux boxes.
Hi,
------- -------
| B |eth0---------eth0| C |
| |eth1---------eth1| |
------- -------
In an attempt to have the install setup to increase speed and/or reliability
of a link between two linux machines (for example in case of a wireless
connection), I read that there were more than one solution, for example the
old eql driver, bonding
2013 Nov 06
1
Re: Calling 'virsh' from within a hook script - avoid deadlock?
Yes.. I saw that, but I have to :)
The issue was solved by converting the hook to Python and using 'subprocess.Popen'. deadlock is gone, all looks good.
Regards
Shlomi
-----Original Message-----
From: Michal Privoznik [mailto:mprivozn@redhat.com]
Sent: Wednesday, November 06, 2013 5:17 PM
To: TSADOK, Shlomi (Shlomi)
Cc: libvirt-users@redhat.com
Subject: Re: [libvirt-users] Calling
2004 Jun 21
4
integrating with existing PBX
I'm looking for a way to give VoIP capabilities to an existing PBX: it's
made by Mitel and it's used in a small/medium environment (a few dozen
phones, but the PBX has capabilities for up to 200, if i remember
correctly).
Any high-level guidelines on how to integrate Asterisk with a PBX that's
already in use? Probably that particular PBX is not supported directly,
but are there
2009 Nov 20
1
What is the relation of CELT and SPEEX?
Hi Jean-Marc,
I got to this mailing list while trying to use the latest SPEEX version
(1.2rc1) VAD capabilities.
I had difficulties to successfully distinguish speech from non speech on
recorded files (probably doing something wrong).
Digging into the code I found out a remark saying :"*The VAD has been
replaced by a hack pending a complete rewrite*".
In search for answer I found out
2006 Jun 13
1
samba 3.0.22 and hebrew file names
Hi
We had an old Sun server running Solaris 2.6 with samba 2.2.2,
Now we upgrade it to Solaris 9 with Samba 3.0.22, but we have one problem.
The file names that are in Hebrew looks on the Windows clients as lines or
squares.
On the old samba server there were no char settings, on the new samba server
I set the char to 862
and the display and unix chars to ISO8859-8 and UTF-8 - it didn't
2007 Dec 20
5
[Bug 13750] New: Repetitively Taking Screenshots Causes the X Server to Hang
http://bugs.freedesktop.org/show_bug.cgi?id=13750
Summary: Repetitively Taking Screenshots Causes the X Server to
Hang
Product: xorg
Version: 7.2
Platform: Other
OS/Version: All
Status: NEW
Severity: normal
Priority: medium
Component: Driver/nVidia (open)
AssignedTo: aplattner
2004 Jul 24
1
Play CD!
MP3s have to use constant bitrate not variable bit rate. Look in the documentation for mpg123.
-----Original Message-----
From: Jozeph Brasil [mailto:jozeph@globalmedia.com.br]
Sent: Saturday, July 24, 2004 5:30 AM
To: asterisk-users@lists.digium.com
Subject: RES: [Asterisk-Users] Play CD!
I do that. But when I play MusicOnHold the music is played slowly! I don?t know why... but is how
2004 Jul 31
0
Trunk doesn't work Adit 600/T100P
Hi !
I am connecting to Adit 600 thru a T100P card
I have configured 1-16 FXS channels and 17-24 FXO.
Everything looks fine on Asterisk side I get a tone
on all FXS channels, but when I try to dialout thru
one of the FXO channels 17-24 it doesn't connect to
the POTS line and echoes back my voice.
I use fxsls and fxols for the T1 channels and ls on
Adit side. Whats wrong here ?
here is my Adit
2004 Jan 17
1
Channel Bank Woes...
Hello all,
I have a new Carrier Access Adit 600 and am having a problem dialing out
and receiving incoming calls.
After the initial ring, the calling party will here nothing. There is a
silence, but the SIP phones that is supposed to ring will ring. The
calling party will not hear the sound of the ring that * sends out on
the line. Additionally, if someone picks up the receiver there is no