similar to: Force TTY

http://bugzilla.mindrot.org/show_bug.cgi?id=829 Summary: Don't allocate a tty if -n option is set Product: Portable OpenSSH Version: 3.8p1 Platform: All OS/Version: All Status: NEW Severity: minor Priority: P2 Component: ssh AssignedTo: openssh-bugs at mindrot.org ReportedBy:

Hi, I'm trying to get OpenSSH to work on Solaris 10 wich Sun C 5.8 compiler (SUNWspro 11). I've compiled OpenSSL 0.9.8a without problem and OpenSSH 4.3p2 as well. [user at compilationserver ~/openssh-4.3p2] ./ssh -V OpenSSH_4.3p2, OpenSSL 0.9.8a 11 Oct 2005 My problem is that I cannot connect to anything. When I try I always get an error [user at compilationserver ~/openssh-4.3p2]

Please Cc: me on the reply as I am not on the list. >From the ssh(1) man page: -t Force pseudo-tty allocation. This can be used to execute arbi trary screen-based programs on a remote machine, which can be very useful, e.g., when implementing menu services. This is similiar to what I am trying to do, use "-t" flag to ssh from my application,

Hello, I've set up a cron job to use ssh with a remote forced command to delete the contents of a directory. System A has the cron job and uses a specific key for system B. No command as such is specified since the key on system B specifies 'command=/bin/rm -rf ...'. However, I am getting error messages back from the cron system (or rather from ssh) about it not allocating a tty since

Hi. >From my understanding the MaxStartups option can be set to limit the number of concurrent sessions the OpenSSH server opens. My concern is how OpenSSH handles the case where this number is reached. >From the code it looks like it simply closes the socket: sshd.c:1440 if (drop_connection(startups) == 1) { debug("drop connection #%d", startups); close(newsock);

https://bugzilla.mindrot.org/show_bug.cgi?id=1995 Damien Miller <djm at mindrot.org> changed: What |Removed |Added ---------------------------------------------------------------------------- Status|NEW |ASSIGNED Assignee|unassigned-bugs at mindrot.org |djm at mindrot.org Attachment #2171|

https://bugzilla.mindrot.org/show_bug.cgi?id=1997 chrysn at fsfe.org changed: What |Removed |Added ---------------------------------------------------------------------------- CC| |chrysn at fsfe.org --- Comment #3 from chrysn at fsfe.org --- Given the approach of distinguishing sockets based on their QoS has not led

** Want to see who you're talking to? Video telephony is growing. A couple of developers has formed the Asterisk Video Task Force in order to improve the support for video telephony in Asterisk for the 1.6 release this fall. There is already support for video in the SIP and the IAX2 channel, but we need to add more in order to improve the support, among other things add video to H.323

Hi, Not sure if this anyone else is interested in this but to me it seems to make sense to use different control session multiplexer sockets for bulk and interactive workloads. Index: auth.c =================================================================== RCS file: /cvs/src/usr.bin/ssh/auth.c,v retrieving revision 1.94 diff -u -p -r1.94 auth.c --- auth.c 23 May 2011 03:33:38 -0000 1.94 +++

Hello ! Like Edmund EVANS reported openssh-2.1.1p4 won't fork to background when using protocol 2. I managed to hack a little patch that might work ... What is the -N command line option supposed to do ? I gather it should work only with protocol2 and without any command to run on the server (and with some port forwardings ??) Anyway in the patch I put some code to check that -N is used

Your log file almost looks like a bug in Asterisk doesn't it? Why call poll() with a zero timeout while passing only one FD? and then why do the read when there is no data? Read the man pages for all the system calls Take a look at the source chan_sip.c /* Wait for sched or io */ res = ast_sched_wait(sched); if ((res < 0) || (res > 1000))

Actually, I have tested this here with an Aastra 9133i and an Asterisk@Home server, and the 9133i will re-subscribe on its own after an Asterisk reboot, if you wait long enough. It took on the order of an hour to do so. Of course, a phone reboot will get it done faster, if necessary, but it _will_ eventually re-subscribe on its own. > Thanks Olle, > > So am I to understand that you

https://bugzilla.mindrot.org/show_bug.cgi?id=1997 --- Comment #4 from Peter Lebbing <peter at digitalbrains.com> --- (In reply to chrysn from comment #3) Sorry for not replying sooner, it slipped my mind! > Would a patch to add a "%I" for "1 for interactive sessions, 0 > otherwise" to the expansion be generally acceptable? Peter, would it > still serve your

Trouble getting chan_sip2 to compile below is what I have done -download acl.c.patch,acl.h.patch,chan2s_sip.c to /root/software cp /root/software/chan_sip2s.c /usr/src/asterisk/channels cd /usr/src/asterisk/ patch -p0 acl.c /root/software/acl.c.patch cd /usr/src/asterisk/include/asterisk patch -p0 acl.h /root/software/acl.h.patch - added the follow to /usr/src/asterisk/channels/Makefile

When creating users in the sip.conf file, they do not appear when running the "sip show users" command from the CLI until i restart. A reload doesnt make them appear. As i said, I am new to the whole Asterisk thing, however have worked with IP/SIP PBX's for a few years - its most likely a user problem though! Check it out and let me know what you get. Cheers Chris PS - I would try

Hi, Ssh -f forks itself in the background. Very usefull if you would like to e.g. tunnel munin over ssh. Now it's tricky to terminate one process if you have multiple running. It seems that ssh currently (looked at 5.1p1) has no write-pid-to-file functionality So I implemented a patch which do so. Tested it a little and it seems to work. Hopefully it is of any use in my form or inspires the

We're proud to announce Astricon 2004 - the first Asterisk user's and developer's conference! * Where? Atlanta, USA * When? September 22-24, 2004 The conference is arranged in partnership with Digium.inc and the keynote speaker is Mark Spencer, lead developer of Asterisk - the Open Source PBX. Among the speakers already signed on are Ed Guy of Pulver.com, John Todd, Jeremy McNamara

OpenSSH 7.1p2 has just been released. It will be available from the mirrors listed at http://www.openssh.com/ shortly. OpenSSH is a 100% complete SSH protocol 2.0 implementation and includes sftp client and server support. OpenSSH also includes transitional support for the legacy SSH 1.3 and 1.5 protocols that may be enabled at compile-time. Once again, we would like to thank the OpenSSH

Friends in the Asterisk.org community, There is a lot of cool stuff going on in Asterisk development, things that will change Asterisk and make it work better in your organisation, make it easier to sell in your area or give you more consulting oppurtunities - in short, functionality that will make a lot of sense for you users. However, developers can't really get anywhere without a

Friends in the Asterisk community, I've been talking for years about the new version of the SIP channel. I've been trying to get funding and get going. Well, the funding part remains to be handled, but I have other news - if you kan keep it to yourself. ...I've began coding. Finally. With a happy smile on my face I removed "pedantic=yes" the other day. After years of