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...classified Product: Portable OpenSSH Version: 5.0p1 Platform: All OS/Version: Linux Status: NEW Severity: normal Priority: P2 Component: ssh AssignedTo: unassigned-bugs at mindrot.org ReportedBy: gst at sysfrog.org When using the ssh command to login to a host, ssh checks if the public key of this host is already known. However, when issuing an IP address instead of a hostname, ssh seems to do a string-based comparison of this IP address with the already known addresses. Example: -------- 8< -------...

...ll use the journal (it did a fsck after the recovering)? why are files corrupted which i don't edit very often (motd, the dpkg list, i changed the resolv.conf before the crashes). i am using ext3-2.4-0.9.6-249.gz cu /gst btw: i'm not subscribed to the list, pls cc replies to me. (gst@sysfrog.org) -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.0.6 (GNU/Linux) Comment: For info see http://www.gnupg.org iD8DBQE7r3LOzsYkmN+2qCgRAqJ/AJ9IY571dWmThAYLzz7BaDlyAhTgGgCeLqh0 APGk1f06FeM+pEcmxAmfs9w= =wzwF -----END PGP SIGNATURE-----

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

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