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...ry for user root tuxx:~ # ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ Rudi Labermeier IT Administrator SP3D Chip Design GmbH Phone: +49 8151 270-119 Petersbrunner Str. 17 Fax: +49 8151 270-200 D-82319 Starnberg mailto:rudi.labermeier@philips.com ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 because of strange reasons my notebook sometimes crashes short after startup (but that's not ext3's fault, maybe mem?, when i wait several minutes it works without problems) the problem is that after 3 crashes at startup, when my notebook finally worked i got the msg: Sep 23 23:29:17 blackbox kernel: EXT3-fs warning (device ide0(3,3)):

...t an idea? Regards, Rudi ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ Rudi Labermeier IT Administrator SP3D Chip Design GmbH Phone: +49 8151 270-119 Petersbrunner Str. 17 Fax: +49 8151 270-200 D-82319 Starnberg mailto:rudi.labermeier@philips.com ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ The information contained in this message is confidential and may be legally privileged. The message is intended solely for the addressee(s). If you are not the inte...

...Do check also >the Settings/Status/Status Messages of your phone for any errors. > >Sascha > >------------------------------------------------------- >Sascha Knific K Systems & Design >Tel. +49-8151-773260 Wittelsbacherstr. 6a >Fax. +49-8151-773262 82319 Starnberg, Germany >Leo +49-8151-773261 WGS84: N57?59,875' E011?20,568' >knific@k-sysdes.net http://www.k-sysdes.net > > >-----Urspr?ngliche Nachricht----- >Von: asterisk-users-admin@lists.digium.com >[mailto:asterisk-users-admin@lists.digium.com] Im Auftrag von tony bank...

Hi, I just wonder, if it is possible, to suppress my own number on outbound dials with chan_capi. I took a look into the sources and think it might work with toggeling the "@" in front of the outbound msn in the dialstring. (Dial(CAPI/@msn... vs. Dial(CAPI/msn... But it doesn't work. Maybee I'm wrong and misunderstood the code. Thanks for any answers! Karsten

...v 09) 01:00.0 VGA compatible controller [0300]: ATI Technologies Inc NI Seymour [AMD Radeon HD 6470M] [1002:6760] wine --version wine-1.3.30 This is with fglrx, I'm trying this now: Code: dmesg | grep fglrx [ 13.298340] fglrx: module license 'Proprietary. (C) 2002 - ATI Technologies, Starnberg, GERMANY' taints kernel. [ 13.341784] [fglrx] Maximum main memory to use for locked dma buffers: 3733 MBytes. [ 13.342076] [fglrx] vendor: 1002 device: 6760 count: 1 [ 13.342497] [fglrx] ioport: bar 4, base 0x5000, size: 0x100 [ 13.342674] [fglrx] Kernel PAT support is enabled [ 13....

https://bugzilla.mindrot.org/show_bug.cgi?id=1489 Summary: ssh should normalize IP addresses before comparison Classification: Unclassified Product: Portable OpenSSH Version: 5.0p1 Platform: All OS/Version: Linux Status: NEW Severity: normal Priority: P2 Component: ssh AssignedTo:

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I think I know the answer to this question based on all of the research I've done, but figured I'd ask anyway. I needed a couple of servers for an HA cluster, and our "order" guy here ordered me a couple of Dell PowerEdge T110 II. I'd planned on using IPMI to fence these things with, but later found out that this model is only one of two PE servers Dell sells that has a

hello, i've an asterisk box which is connected to an E1/PRI via a TE110P card. incoming calls from mobile phones where the number is transfered as a whole block work fine, but when dialing from an analog or ISDN line to the asterisk box there is a timeout of about 3-5 seconds. originally my incoming context looked like: exten => _X.,1,Dial(SIP/${EXTEN}@domain.tld) so i assumed that the

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031207/e5e9b8eb/attachment.htm -------------- next part -------------- Hello all, I am newbie to Telephony world (IP and PSTN). Please excuse me if you find my questions very dumb. I am trying to configure my IP 7940 with the Asterisk, when phone boots up it only shows the message "Phone

Hi all, I compiled/installed chan_capi.so without problems. When I launch Asterisk, I get the following error: --- [chan_capi.so] => (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found Mar 30 19:47:52 NOTICE[16384]: chan_capi.c:2338 mkif: ast_capi_pvt(91xxxxxx,*,pstn,0x2,2) (1,2,64) (0)(0.800000/0.800000) 0 Mar 30 19:47:52 NOTICE[16384]: chan_capi.c:2338

Hello Everyone, This morning I received a personal email from a recent poster asking "Why is Mr. Eastep so mean?". The post went on to explain that I had responded abruptly to the person''s request for help and asked why I wasn''t more helpful to people attempting to understand something new. Good question. Over the last nine months, I have been struggling to deal with

It is with regret that I announce that Shorewall development and support is officially ended. Sean''s post has finally driven it home to me that in the long term, trying to support a project like Shorewall is impossible for a person of my personality and age. Sean -- please believe that this isn''t about you or your post -- your post was just the proverbial straw on this old

On Sun, Nov 07 2010 at 6:05pm -0500, Andi Kleen <andi@firstfloor.org> wrote: > On Sun, Nov 07, 2010 at 10:39:23PM +0100, Milan Broz wrote: > > On 11/07/2010 08:45 PM, Andi Kleen wrote: > > >> I read about barrier-problems and data getting to the partition when > > >> using dm-crypt and several layers so I don''t know if that could be > >

hello, i'm currently using asterisk (cvs-head) as PSTN gateway. the routing logic is mostly done in OpenSER. the problem is that i'm not able to transfer calls between the PSTN and another SIP peer (when the PSTN<>SIP connection goes over asterisk but the SIP<>SIP connection does not). there are 2 possibilities for asterisk to be part of the transfer: 1) asterisk receives

Hello, I'm currently using OpenSER as REGISTER server and Asterisk for the call routing. Do i need the OpenSER nathelper module if i want to offer (mostly) automatic NAT traversal to my users or does Asterisk have the same functionality? It seems that the nathelper module should be able to automatically traverse any NAT as long as the User-Agents use symmetric RTP. Further it is possible (in