search for: darrick

...rovides > asynchronous host page cache flush mechanism. We don't > support 'MAP_SYNC' with virtio pmem and xfs. > > Signed-off-by: Pankaj Gupta <pagupta at redhat.com> > --- > fs/xfs/xfs_file.c | 9 ++++++--- > 1 file changed, 6 insertions(+), 3 deletions(-) Darrick, does this look ok to take through the nvdimm tree? > > diff --git a/fs/xfs/xfs_file.c b/fs/xfs/xfs_file.c > index a7ceae90110e..f17652cca5ff 100644 > --- a/fs/xfs/xfs_file.c > +++ b/fs/xfs/xfs_file.c > @@ -1203,11 +1203,14 @@ xfs_file_mmap( > struct file *filp, &g...

...rovides > asynchronous host page cache flush mechanism. We don't > support 'MAP_SYNC' with virtio pmem and xfs. > > Signed-off-by: Pankaj Gupta <pagupta at redhat.com> > --- > fs/xfs/xfs_file.c | 9 ++++++--- > 1 file changed, 6 insertions(+), 3 deletions(-) Darrick, does this look ok to take through the nvdimm tree? > > diff --git a/fs/xfs/xfs_file.c b/fs/xfs/xfs_file.c > index a7ceae90110e..f17652cca5ff 100644 > --- a/fs/xfs/xfs_file.c > +++ b/fs/xfs/xfs_file.c > @@ -1203,11 +1203,14 @@ xfs_file_mmap( > struct file *filp, &g...

...situation (we're doing some testing before switching completely to an Asterisk based system), is there something I can try to improve the percentage of calls that are disconnected? I would use silence detection in voicemail.conf but wanted to know if there was some other method. Thanks, Darrick -- Darrick Hartman DJH Solutions, LLC 877.901.3113

Hi all, This patchset makes some key modifications to the original ''stable page writes'' patchset. First, it provides users (devices and filesystems) of a backing_dev_info the ability to declare whether or not it is necessary to ensure that page contents cannot change during writeout, whereas the current code assumes that this is true. Second, it relaxes the

...he log code holds one reference. The end result is the warning, but the truth is that we set the flag EXTENT_FLAG_LOGGING only during fsync. So clear flag EXTENT_FLAG_LOGGING for extent maps split from a large one. Reported-by: Johannes Hirte <johannes.hirte@fem.tu-ilmenau.de> Reported-by: Darrick J. Wong <darrick.wong@oracle.com> Signed-off-by: Liu Bo <bo.li.liu@oracle.com> --- fs/btrfs/file.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/fs/btrfs/file.c b/fs/btrfs/file.c index 83c790d..7bdb47f 100644 --- a/fs/btrfs/file.c +++ b/fs/btrfs/file.c @@ -59...

Hi All. I have notice a problem with FM feature (screen macros) on Asterisk CVS version. When call goes via IAX and calling part "accept the call" on Dial command with option M, in macros context it's setting MACRO_RESULT=CONTINUE, but anyway it hangups both channels. If anyone faced with such problem please let me know. I need to know whether it's bug or just configuration

...e flush mechanism. We don't > > support 'MAP_SYNC' with virtio pmem and xfs. > > > > Signed-off-by: Pankaj Gupta <pagupta at redhat.com> > > --- > > fs/xfs/xfs_file.c | 9 ++++++--- > > 1 file changed, 6 insertions(+), 3 deletions(-) > > Darrick, does this look ok to take through the nvdimm tree? <urk> forgot about this, sorry. :/ > > > > diff --git a/fs/xfs/xfs_file.c b/fs/xfs/xfs_file.c > > index a7ceae90110e..f17652cca5ff 100644 > > --- a/fs/xfs/xfs_file.c > > +++ b/fs/xfs/xfs_file.c > > @@ -1...

Is it possible with asterisk to setup aliases for SIP? For example, direct sales@mysipdomain.com to 55544@mysipdomain.com If this isn't possible directly with asterisk, does SER offer anything along those lines? A search of the usual sites didn't turn up anything conclusive. Thanks, Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com

Digium has released version 3.0.3 of its product registration utility. This is the first version of the registration utility that is compiled against the uClibc C library. A benefit of this transition is that the register binary should run more consistently and reliably across a wider range of Linux distributions. The new versions of 'register' and 'asthostid' can be

...linux-fsdevel at vger.kernel.org S: Supported F: src/vfs/ @@ -163,6 +167,7 @@ S: Supported F: tests/ocfs2/ OVERLAYFS +R: Amir Goldstein <amir73il at gmail.com> L: linux-unionfs at vger.kernel.org S: Supported F: tests/overlay @@ -174,6 +179,7 @@ S: Supported F: tests/udf/ XFS +R: Darrick J. Wong <djwong at kernel.org> L: linux-xfs at vger.kernel.org S: Supported F: common/dump -- 2.39.2

I havent got any mails since 2:42 this morning..usually i get at least the normal 10-15 a hour, if someone gets this can they reply? Thanks! Ron -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060308/c04e3fc9/attachment.htm

Hi Everyone, Knowing that running Asterisk on an embedded board like the Alix2d3 requires some fine tuning. Do you know of any good guides out there that does this from beginning to end? Looking to run this in a small office environment. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Darrick J. Wong would like to nominate Anand Jain to help more on btrfs testing part (tests/btrfs and common/btrfs). He would like to be a co-maintainer of btrfs part, will help to review and test fstests btrfs related patches, and I might merge from him if there's big patchset. So CC him besides send...

...It appears they have a 'feature' which cuts out the incoming audio if a loud noise (simulated by blowing into the receiver) is experienced outgoing. Pretty much going to a true analog service is the only solution that I can think of. Would be interested if anyone has other thoughts. Darrick

Using a Polycom 550 and 650 phones on my Asterisk server for testing. I can't figure out why the volume is so low. How can I adjust the volume control on Asterisk? It's at max on the handset phones. Thanks! Hin

Hi there, Here is my dilema. I have a new small business customer that wants me to put in a VoIP phone system for them. Based on their requirements, I have determined that it needs to be a "set it and forget it" type of thing like a lot of small business proprietary systems. At the same time they would like to be able to do minor dial plan changes themselves so I have determine

Hello! I'm trying to get icecast working in windows with ezstream...i know it should be easy enough to figure out lol..i'm new to icecast. i get an error from ezstream...cannot open playlist.m3u...I've made a text file called playlist.m3u and put it in the web directory of icecast...is this the right location? what do i have to put into the playlist to get it up and running? Thank

We currently have most of our voicemail forwarded to user's email addresses, but the message is coming in at a way low volume. It sounds great when you listen on the phone, but it's very hard to hear when you listen on the computer. Does anyone know of a way to increase the gain on the file before sending it off? Aaron

Hi everyone, I am trying to get iaxmodem up and running. This is a very basic setup, which at this moment should only answer incoming faxes. What I did: zapata.conf (rest of it should be fine): faxdetect=incoming group = 1 channel => 1-2 context=from-pstn iax.conf: [200] username=200 type=friend callerid="Fax" <200> secret=dooo host=dynamic notransfer=yes allow=all

Hi, I have to install an Asterisk PBX for a customer and he wants something like logic supply's fanless computers. Can anybody advise about how good will they work, are they compatible with the Asterisk system? I'll also be installing a sangoma 4 port FXO card in it. -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was scrubbed... URL: