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  • Mac internal mic not working with Xlite

    - by Pablo Fernandez
    I've used xlite for a while now but all of the sudden the internal mic from my mac stopped to work with it. It does call, and you can see the red bar rising but the other person can't hear a thing. The mic does work when using Skype so I think it's a problem with Xlite Things I've already tried: Reconfigure Xlite Reinstall Xlite Install Xlite Beta (After every step I rebooted the macbook) Obviously none of them worked. Thanks

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  • Is it better to cut and store all sprites needed from a spritesheet in memory, or cut them out just-in-time?

    - by xLite
    I'm not sure what's best practice here as I have little experience with this. Essentially what I am asking is... if it's better to get your single PNG with all your different sprites on it for use in-game, cut out every sprite on startup and store them in memory, then access the already-cut-out sprite from memory quickly or Only have the single PNG with all the different sprites residing in memory, and when you need, for example, a tree. You cut out the tree from the PNG and then continue to use it as normal. I imagine the former is more CPU friendly than the latter but less memory friendly, vice versa for the latter. I want to know what the norm is for game dev. This is a pixel based game using 2D art. Each PNG is actually an avatar's sprite sheet with each body part separated and then later joined to form the full body of the avatar.

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  • Unable to call through asterisk

    - by sk
    I want to create a voip service. I have installed asterisk-1.4 on a dedicated remotely hosted debian lenny distro. I made a sip.conf and extensions.conf so as to place a call between two sip phones(i am using xlite 3.0) installed in some other Windows PC. Whenever i switch this phones the asterisk console shows that Registration from '"1000"<sip:[email protected]>' failed for '122.168.10.254' - Peer is not supposed to register Where xx.xx.xx.xx is the server's IP. i.e my sip phones are unable to register with the asterisk server. Please help me to place call between two sip phones #sip show peers Name/username Host Dyn Nat ACL Port Status 2000 (Unspecified) D 0 Unmonitored 1000 (Unspecified) D 0 Unmonitored 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline] # sip show registry Host Username Refresh State Reg.Time # sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message 0 active SIP channels

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