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• How do I create midi data to trigger an Ultrabeat pattern in Logic?

  - by user289581

  I've made some Ultrabeat patterns but I can't figure out how to trigger them. I make the pattern on C-1, then I go back to the arrange window and hit record and then hit C1 on my keyboard, and it doesn't play the pattern, it just plays a single note. I have Pattern Mode enabled on Ultrabeat and I'm using a one-shot trigger but I can't seem to trigger my pattern to play at all. What am I doing wrong?

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• SSH traffic over openvpn freezes under weird circumstances

  - by user289581

  I have an openvpn (version 2.1_rc15 at both ends) connection setup between two gentoo boxes using shared keys. it works fine for the most part. I use mysql, http, ftp, scp over the vpn with no problems. But when I ssh from the client to the server over the vpn, weird things happen. I can login, i can execute some commands. But if i try to run an ncurses application like top, or i try to cat a file, the connection will stall and I'll have to sever the ssh session. I can, for example, execute "echo blah; echo .; echo blah" and it will output the three lines of text over the ssh session fine. But if i execute "cat /etc/motd" the session will freeze the moment I press enter. While it seems like a terminal emulation problem it makes no sense why using the vpn would affect the ability for ssh to render things correctly. I am at a loss to explain why everything else works, including scp, but ssh just breaks over the vpn. Any thoughts ?

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• Asterisk + FreePBX + GoTalk. Inbound route not working.

  - by user289581

  I'm running asterisk 1.6.2.6 and freepbx-2.7.0 My trunk is configured as follows: Outgoing Settings Trunk name: GoTalk Peer Details: host=sip.gotalk.com username=09xxxxxx secret=YNxxxxxx type=peer fromuser=09xxxxxx fromdomain=sip.gotalk.com canreinvite=no insecure=very Incoming Settings User Context: 09xxxxx User Details: username=09xxxxx fromuser=09xxxxx type=peer secret=YNxxxxx insecure=very host=dynamic fromdomain=sip.gotalk.com context=from-pstn Register String: 09xxxxxx:[email protected]/09xxxxxx I have an inbound route called Incoming with DID 09xxxxxx diverted to local extension 200 When I do a sip trace and dial my telephone number 0741xxxxx I just get failure beeps. I never see any SIP traffic from GoTalk to my asterisk server trying to connect the call. Seems I'm not registering correctly for incoming calls because GoTalk aren't sending them to me. I am correct in using the GoTalk username 09xxxxxx as the DID, aren't I ? I've tried using my phone number but it makes no difference.

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