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Dear all, > Caveats > ======= > Features which OCFS2 does not support yet: > - extended attributes > - readonly mount > - shared writeable mmap > - loopback is supported, but data written will not > be cluster coherent. > - quotas > - cluster aware flock <---------------------------------------- >

V1 -> V2: Move kmalloc() before mutex_lock(), suggested by Arjan. file_remove_suid() should be called with i_mutex held, file_update_time() too. So move them after mutex_lock(). Plus, check the return value of kmalloc(). Signed-off-by: WANG Cong <amwang@redhat.com> Cc: Arjan <arjan@infradead.org> Cc: Chris Mason <chris.mason@oracle.com> Cc: Yan Zheng

Hello, I'm using X-web lite in a webpage to connect to one of our asterisk server. But now I have a problem, when you are connected to a voice script the voice will not be heard after a couple of seconds. When you press or say something that the voice will come back for a couple of seconds. When I thy X-Lite (stand-alone version) I had the same problem, but when I turned off the

start it with asterisk -vvvgc bkw ----- Original Message ----- From: "Arjan" <arjan@inventionz.org> To: <asterisk-users@lists.digium.com> Sent: Sunday, July 11, 2004 12:27 PM Subject: [Asterisk-Users] Asterisk on FreeBSD 4.10 dies > Hi All, > > I'm pretty green to Asterisk. I'm trying to work towards a basic setup > with a couple of Cisco 7960's

V2 -> V3: set ''err'' to -ENOMEM when kmalloc() fails. Thanks to Tao. V1 -> V2: Move kmalloc() before mutex_lock(), suggested by Arjan. file_remove_suid() should be called with i_mutex held, file_update_time() too. So move them after mutex_lock(). Plus, check the return value of kmalloc(). Signed-off-by: WANG Cong <amwang@redhat.com> Cc: Arjan

Hi, Does anybody know if RAMI (Ruby Ami) is still functional? And is this still compatible with asterisk 1.8 Best Regards, Arjan Kroon Mobillion BV

Arjan, I am forwarding your email to the nut-upsuser mailing list. I am not an expert on the Powermust driver; hopefully somebody on the mailing list will be able to help you! -- Peter Arjan DJ wrote: > > Dear Peter, > Some days ago I bought a Sweex UPS (500VA) and now I have set up NUT (using > the Powermust driver) and everything is working fine. Invoking upsmon -c fsd > gives

Hi all, I have installed an asterisk server at my office, the server is behind a firewall. On the firewall I've set NAT a rule for incoming traffic on port 5060 to be forwarded to the server. Connecting from home with my sip client doesn't work at all. The asterisk server itself is ok, when I make a local connection at my office, 10.0.0.129 (client) to 10.0.0.6 (asterisk server) it

Hello, I'm a newbie in connection several asterisk servers with each others. I've got the following situation. I've got 9 asterisk servers (asterisk00 till asterisk08). When I call to asterisk08 then I want to redirect an application which runs on asterisk00. But how can I redirect in an application on asterisk08 to an application on asterisk00? Or isn't this possible?

Hi, In my application I want to have the sysdate + 5 minutes. I know that the sysdate is in the variable ${DATTIME} But now I want to now how I get the sysdate + 5 minutes into a variable? Doe's anybody knows the answer? Kind Regards Arjan Kroon -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, How do you all handle the situation when a centrale fastagi server process(es) are down? AGI(..) prints "Unable to locate host" and the dailplan jumps to extension h. I'd like to handle the return value and keeping the caller in the dailplan and not to the hangup extension. Any tips about how to handle a AGI(..) returns -1 condition? thx Arjan Kroon Mobillion BV

Hi, We encounter a problem with the variable CALLERID(dnid) We use E1 lines where we can make an inbound call or an outbound call on the same channel (not at the same time) If the CALLERID(dnid) is not used, than the CALLERID(dnid) will be the CALLERID(dnid) of the previous call For example: - First we get a inbound call on channel DAHDI/11-1 with CALLERID(dnid) = '655871460' We read

Hai, Does anybody have problems with a wrong Connected Line ID with asterisk version 1.6 The following bug was for version 1.4, but I cannot make up if this bug is still in version 1.6 http://forums.digium.com/viewtopic.php?t=7780 In version 1.8 it is possible to change the Connected Line ID, but this isn't the case in version 1.6 Regards, Arjan Kroon Mobillion BV

Hi, I'm using the Read command the read a DTMF tone. In this read command I play a voice-file. But now when I press one off they keys of my telephone the voice-file will stop playing a the program go the next priority. Is it possible to play the voice-file until the right DTMF tone is pressed? (say for instance the Zero). Kind regards Arjan Kroon Mobillion B.V.

On Tue, Feb 6, 2018 at 4:46 PM David Woodhouse <dwmw2 at infradead.org> wrote: > On Wed, 2018-02-07 at 00:36 +0000, Chandler Carruth wrote: > > > > > > > That would be __x86_indirect_thunk but the kernel doesn't use it. > > > We use -mindirect-branch-register and only ever expect the compiler > > > to use the register versions which are

Hi, Can anybody tell me which version of res_perl I have to install on Asterisk 1.2.4. I tried to compile res_perl version 3.5 on Asterisk 1.2.4 and I got the following error. gcc -Wall -DRES_PERL_BASE="\"/usr/local/res_perl\"" -DMULTIPLICITY - D_REENTRANT -D_GNU_SOURCE -DTHREADS_HAVE_PIDS -fno-strict-aliasing -pipe -Wdeclaration-after-statement

Hi, We are using asterisk version 1.0.0. For queue'ing we use the rawplayer script to play a music file in the background. Now we see that after a while all the sessions on our Linux environment will be taken by the rawplayer process. An example of such a session is (done with ps -ax|grep rawplayer) 24785 ? Z 0:00 [rawplayer <defunct>] 8415 ? Z

Hi, Is it possible to use a live audio stream in asterisk I want to call a number and then hear an external audio stream. For example http://www.radioveronica.nl/radioveronicaplayer/radioveronica.asx I thought it was possible to use musiconhold, but I did not get it working. This is my musiconhold.conf ; ; Music on Hold -- Sample Configuration ; [general] [default] mode=custum

Hi all, I&#180;ve been developing a search application using Xapian and Omega. When our visitors search on specific keywords it&#180;s noticable that Omega is case sensitive. It will find results on keyword `Asus&#180; and it will NOT find results on keyword &#180;asus&#180;. I&#180;ve been looking through the documentation to find a CGI param to turn this

1.) after initializing a ogg_stream_state may I just keep calling ogg_stream_packetin with valid ogg_packet's until no more ogg_packets needs to be added? Or do I need to call ogg_stream_pageout after every ogg_stream_packetin? Iaw may I delay calling ogg_stream_pageout until there are no more packets to be added? 2.) I assume I have call ogg_stream_pageout until no more pages can be