similar to: [Bug 1282] Log which key used for authentication

http://bugzilla.mindrot.org/show_bug.cgi?id=1282 Summary: Log which key used for authentication Product: Portable OpenSSH Version: -current Platform: All OS/Version: All Status: NEW Keywords: low-hanging-fruit Severity: security Priority: P2 Component: sshd AssignedTo: bitbucket at

https://bugzilla.netfilter.org/show_bug.cgi?id=1282 Bug ID: 1282 Summary: SIGSEGV on loading tables Product: nftables Version: unspecified Hardware: x86_64 OS: Ubuntu Status: NEW Severity: normal Priority: P5 Component: nft Assignee: pablo at netfilter.org Reporter:

https://bugzilla.mindrot.org/show_bug.cgi?id=2666 Bug ID: 2666 Summary: Ability to specify minimum RSA key size for user keys Product: Portable OpenSSH Version: -current Hardware: All OS: All Status: NEW Severity: enhancement Priority: P5 Component: sshd Assignee:

Howdy, I'm setting up several gluster 3.12 clusters running on CentOS 7 and have having issues with glusterd.log and glustershd.log both being filled with errors relating to null client errors and client-callback functions. They seem to be related to high CPU usage across the nodes although I don't have a way of confirming that (suggestions welcomed!). in

Hello using Asterisk 1.8.32.3. What is the best way of knowing a call is being transfered (attended and unattended) ? And also knowing whereto (sip user) the call is being transfered and who is the transferer ? So I can log this information. Kind regards. J. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Here are some notes/observations I've done on my setup that works for me. Hopefully it will be added to some r-forge documentation/wiki. I use: Windows Vista Business SP2 32bit TortoiseSVN 1.6.7 (Build 18415 - 32 Bit , 2010/01/22 17:55:06) PuTTY v0.60 (with Pageant v0.60) I have a deprecated r-forge project ('r-oo') that I will use through out. My r-forge username is

Thanks Milind, Yes I?m hanging out for CentOS?s Storage / Gluster SIG to release the packages for 3.12.1, I can see the packages were built a week ago but they?re still not on the repo :( -- Sam > On 18 Sep 2017, at 9:57 pm, Milind Changire <mchangir at redhat.com> wrote: > > Sam, > You might want to give glusterfs-3.12.1 a try instead. > > > >> On Fri, Sep

I have been working through the book "Applied longitudinal data analysis: modeling change and event occurrence" by Judith D. Singer and John B. Willett. I have been working examples using SAS and also using it as an opportunity for learning to use R for statistical analysis. I ran into some difficulties in chapter 8 which deals with using structural equation modeling. I have tried to

Sam, You might want to give glusterfs-3.12.1 a try instead. On Fri, Sep 15, 2017 at 6:42 AM, Sam McLeod <mailinglists at smcleod.net> wrote: > Howdy, > > I'm setting up several gluster 3.12 clusters running on CentOS 7 and have > having issues with glusterd.log and glustershd.log both being filled with > errors relating to null client errors and client-callback

We're using Asterisk 14.7.6 and I have a dialplan that ends like this: same => n,Dial(SIP/${EXTEN:0:4}@peer1) same => n,Set(GLOBAL(EpochAtCallEnd)=${EPOCH}) same => n,Hangup() When peer1 hangsup, the priorities after the Dial are executed fine. But when the caller hangsup during the Dial, the cleanup steps aren't done. Why? I did read "Note that on a successful

Hi, asterisk admin and users. I need to SIP INVITE uri with domain via peer. And uri domain differ then peer domain. dialplan: exten => s,n,Dial(SIP/peer1/number at domain2.com,60,r) [peer1] type=friend host=domain1.com fromdomain=domain1.com As a result in SIP packet uri: number at domain2.com@domain1.com I need: number at domain2.com I can't use "SIP uri dial", i need

Hi, I have installed asterisk-1.8.25.0 on an Ubuntu server which has both an ipv6 ip and ipv4 ip(real ip) assigned. And I have a client ubuntu with only ipv4 ip(local ip) installed asterisk-1.8.25.0 . I want to configure the client asterisk with the server asterisk as IAX2 peer and want to connect to the IPV6 ip. I bind the server with ipv6 and also sending the registration request from the

The following is a pxelinux problem, specifically to do with including config files with the menu include directive and the ifcpu64.c32 com module. I have a working ifcpu64.c32 setup that jumps to the label rescue64 in the case of a 64-bit CPU. The label "rescue64" defines a 64-bit kernel and a 64-bit initrd.img. The setup jumps to a label named "rescue32" in the case of a

FYI - I've been testing the Gluster 3.12.1 packages with the help of the SIG maintainer and I can confirm that the logs are no longer being filled with NFS or null client errors after the upgrade. -- Sam McLeod @s_mcleod https://smcleod.net > On 18 Sep 2017, at 10:14 pm, Sam McLeod <mailinglists at smcleod.net> wrote: > > Thanks Milind, > > Yes I?m hanging out for

Hello I have two peers setup and working with x2 bricks each. They have been working via tcp for the last 4-5 months. I just got two Infiniband cards and put the on the peers. I want to change the transport type to rdma instead of tcp but I don't see an easy way to do this. Can you please help me with proper instructions. Best Regards Ivan Dimitrov

I have considered opening a bug report on this, but wanted to get some feedback and make sure I am not missing something in the way of a simple work around. What is the scenario in which this impacts your implementation? Ours is the desire to use the same realtime SIP database for many asterisk servers, and route the call based on a "home server" value in the realtime database. The

Hello everybody I’m french and ‘im a network administrator. It the firs time I write on this mailing list .. I configure a debian distribution (the last one sarge) with iproute2 to route packets depending on source ip address It works fine but it is very very slow . When I make a ping for example on a windows machine : [in French ] $> ping www.Google.Fr Envoi

Thanks, Anthony. I added both 'g' and 'F' options. Now, when the caller hangs-up, my cleanup code is run by both the caller channel and the peer channel, but I only want the caller channel to do that. Also, when the peer hangs-up, there is no execution of the priorities following the Dial. Finally, is there a way to reset all globals, maybe as a variant of "dialplan

Hello list, I try to setup an asterisk-server with different SIP-Peers to PSTN. The Peer are working and configured in sip.conf: [peer1] type=peer host=10.10.10.1 [peer2] type=peer host=10.10.10.2 Now dialout is no problem. Extensions.conf says: exten => _0Z.,1,Dial(SIP/49${EXTEN:1}@peer1,30) But how can I setup a failure-route if the SIP-Proxy "peer1" ist not

Hi, We are preparing to make the release of OpenSSH 4.8 soon, so we would greatly appreciate testing of snapshot releases in as many environments and on as many operating systems as possible. The highlights of this release are: * Added chroot(2) support for sshd(8), controlled by a new option "ChrootDirectory". Please refer to sshd_config(5) for details, and please use this