search for: boedick

...ion and calls another party and connects the two. Is there a way using the Manager API to check back later on the status of this call (is it still up, etc.)? I have found the "Status" API action, but I don't know how to get what to pass in for the channel parameter. Thanks, Matthew Boedicker

ssh-agent does not allow you to remove individual keys without having the key file that was added. To remove these keys the user must remove all keys with ssh-add -D. Would a patch to make ssh-add skip the existence check for the file be considered? The specific use case is that a USB drive is mounted with the key, the key is added to the agent then the USB drive is unmounted.

...ringback in either case. I have also tried variations of the progressinband setting. I have listened to the RTP going from the SIP server to Asterisk and I can hear the ringing in it. It seems like Asterisk isn't sending any audio to the first caller until both parties answer. Thanks, Matthew Boedicker

...Product: Portable OpenSSH Version: 5.5p1 Platform: All OS/Version: All Status: NEW Severity: normal Priority: P2 Component: Documentation AssignedTo: unassigned-bugs at mindrot.org ReportedBy: matthewm at boedicker.org The "-c cipher_sec" section says that 3des is still the default cipher. Also newer ciphers like aes and arcfour are not listed. -- Configure bugmail: https://bugzilla.mindrot.org/userprefs.cgi?tab=email ------- You are receiving this mail because: ------- You are watching the as...

I have an application that uses the Asterisk Management Interface to bridge two calls using the Originate command with Dial as the action. Using one SIP server, there is no ringback on the second leg of the call. The first person is called, answers, and hears silence until the second person picks up, even though the second person's phone is ringing. When the call goes to another SIP gateway,