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| 01:33.25 | HarryR`Windows | ast_rtp_bridge() and friends should allow me to do a Dial() and have audio pass through from the calling party to the destination without any transcoding in asterisk right? |
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| 12:40.10 | *** mode/#asterisk-dev [+oooo bmd bkruse Deeewayne putnopvut] by irc.freenode.net |
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| 14:30.01 | dklima | msetim: oie afim d tc? |
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| 19:04.52 | *** join/#asterisk-dev SirWhit (n=sirjames@blk-222-38-6.eastlink.ca) |
| 19:05.18 | SirWhit | hi there |
| 19:07.32 | blitzrage | howdy |
| 19:07.52 | SirWhit | hey blitzrage! |
| 19:07.59 | SirWhit | no too many folks on today eh |
| 19:08.15 | blitzrage | its sunday... :) |
| 19:08.20 | SirWhit | hehehe true enough |
| 19:08.29 | SirWhit | are you a asterisk-code guru? |
| 19:08.38 | blitzrage | sorry, can't say I am |
| 19:08.51 | SirWhit | no worries.. me either |
| 19:09.03 | SirWhit | just trying a couple things out and need some direction |
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| 23:31.04 | *** mode/#asterisk-dev [+o Qwell] by ChanServ |
| 23:31.29 | file | Qwell: go away! |
| 23:32.22 | Qwell | bip died O.o |
| 23:32.35 | file | eep |