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...and an authorized_keys file exists, checking the existing authorized_keys file takes precedence. I believe this to be a more simplistic and trivial patch to openssh opposed to the pre-existing patches, such as the popular LDAP patch (http://code.google.com/p/openssh-lpk/). Best, -- Michael J. Flickinger -------------- next part -------------- A non-text attachment was scrubbed... Name: patch.diff Type: text/x-patch Size: 5655 bytes Desc: not available URL: <http://lists.mindrot.org/pipermail/openssh-unix-dev/attachments/20111214/437dbeba/attachment.bin>

Hello, It is my understanding that on the softphone side, asterisk is only responsible for establishing the session between two phones. If this is the case, does it matter what type of audio codecs the two phones are using? And if it does matter, are there any codecs that cause problems with asterisk bridging two SIP connections? Thanks for your helpful input, Daniel

https://bugzilla.mindrot.org/show_bug.cgi?id=1663 Petr Lautrbach <plautrba at redhat.com> changed: What |Removed |Added ---------------------------------------------------------------------------- CC| |plautrba at redhat.com -- You are receiving this mail because: You are watching the assignee of the bug. You

Hello, I'm new to Asterisk, and am trying to get the basic features under my belt until I move on to the more advanced ones. Currently I have two softphones registered with my * server on my network, and one of the phones can call the other just fine, but when I try to call from the other phone, I get a message on the phone saying that the call was terminated for reasons unknown.

I'm trying to get two linux machines with kphone-3.11 two communicate with each other over asterisk. I have them configured correctly on asterisk to use sip channels, but when I call from one phone to the other I don't any voice communication between the phones. According to the phones I'm connected, but according to asterisk, I get the following message: -- Executing

Moshe, I was having the same problem with my software only asterisk pbx setup. I was using two kphones on different machines, connecting through a machine running asterisk. They would connect just fine, but voice was not getting routed through. I installed linphone, which can be found at wwww.linphone.org, on the machines running kphone, and can now get very clear voice communications,