search for: tsol

I try to run tcptop in solaris 10 x86 with patch level 125101-04 and I get the following error /usr/include/sys/zone.h line 16: can''t find include file sys/tsol/label.h error. Is there a work around for this problem? -- This message posted from opensolaris.org

...stem_labeled 6293271 Zone processes should use zone_kcred instead of kcred 6394554 integrate Solaris Trusted Extensions Files: create: deleted_files/usr/src/cmd/dminfo/Makefile create: usr/src/cmd/allocate/dminfo.c create: usr/src/cmd/devfsadm/devalloc.c create: usr/src/cmd/ldap/ns_ldap/ldapaddtsol.c create: usr/src/cmd/lp/lib/lp/tx.c create: usr/src/cmd/mdb/common/modules/genunix/tsol.c create: usr/src/cmd/mdb/common/modules/genunix/tsol.h create: usr/src/common/tsol/blabel.c create: usr/src/lib/libbsm/common/devalloc.c create: usr/src/lib/libbsm/common/devalloc.h create: usr/src/lib/...

...ww.sun.com/msg/ZFS-8000-8A scrub: scrub in progress for 0h1m, 22.57% done, 0h5m to go config: NAME STATE READ WRITE CKSUM rpool DEGRADED 0 0 17 c0d0s0 DEGRADED 0 0 34 too many errors errors: Permanent errors have been detected in the following files: //dev/.devfsadm_dev.lock //var/svc/log/system-tsol-zones:default.log //var/svc/log/system-labeld:default.log //var/svc/log/system-filesystem-volfs:default.log Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://mail.opensolaris.org/pipermail/zfs-discuss/attachments/20100630/b035c9a0/attachment.html>

Hi, I am getting this problem when trying to register with Voipfone.co.uk. It used to work, and I haven't changed anything that I know of. Jan 13 10:22:37 WARNING[21645]: acl.c:213 ast_get_ip_or_srv: Unable to lookup 'voipfone.co.uk.voipfone.co.uk' Why does the domain name appear twice? I don't know when it stopped working. In SIP.CONF [sip_proxy-out] type=peer

...------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050118/1d3c83 e1/attachment-0001.htm ------------------------------ Message: 11 Date: Tue, 18 Jan 2005 10:50:30 +0200 From: "David Norton" <asterisk@tsol.co.za> Subject: RE: [Asterisk-Users] Best Grandstream firmware to use? To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com> Message-ID: <E1Cqp4u-00033e-00@guinness.msol.co.za> Content-Type: text/plain; charset="us-a...

...hink of is 42. Thanks in advance, T. -- Theodore=J.=Soldatos=_\_="There=is=always=a=bug=somewhere",=said==HAL=to=the== = theodore@eexi.gr =_/_==Ultimate=Programmer,=and=turned=off=the=air=supply.= = bafh@hellug.gr =_\_="Everybody=knows=the=war=is=over,==================== = tsol@space.gr =_/_==everybody=knows=the=good=guys=lost"===Leonard=Cohen= =====================_\_============ http://w4u.eexi.gr/~theodore ============ === Space Hellas ====_/_=========== Finger: theodore@aurora.eexi.gr ========== Return-Path: <antoon@cas.et.tudelft.nl> Delivered-To: samba...

Sorry about that, I was been an idiot again! I had openssl and libssl installed, but had the wrong version of libssl-dev. _____ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of David Norton Sent: Tuesday, December 28, 2004 10:02 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users]

Hi, I have several Grandstream phones connected to Asterisk, some behind NAT and others not. If I reboot all the phones, everything is fine. Should the connection go down, and then come back again, those behind a NAT are still able to make calls, but are unable to receive calls. -- Executing Dial("SIP/1239-ba74", "SIP/1242|60|t") in new stack Jan 12 23:45:19

Hi, Currently I only have 1 PRI which I am using for dial-in customers. The line is connected to a Portmaster3. I have never used more than 10 concurrent channels. The calls can be both analog or ISDN. It would be a waste to order another PRI for my Asterisk box. Is there any way of splitting a PRI into 2 PRI's of 15 channels each, or plugging the PRI into the * box and it send the data

Hi All, I have the following scenario, it may already have been answered elsewhere, but I cant find the solution. I already have a PBX and would like to start implementing asterisk. I have ordered a 4 port card from the asterisk store (2 port FXS and 2 port FXO) and am waiting for it to arrive. I do not want to plug my incoming lines into my FXO ports yet as not all the desks have IP phones