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[asterisk-dev] [Code Review] New signaling module to handle

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jpeeler at digium.com
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Posted: Tue May 26, 2009 5:10 pm Post subject: [asterisk-dev] [Code Review] New signaling module to handle

----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: http://reviewboard.digium.com/r/253/ ----------------------------------------------------------- (Updated 2009-05-26 12:58:44.830116) Review request for Asterisk Developers. Changes ------- Fixed a problem with three way calling. Changed linking for sig_analog.o and updated the copyright headers. Summary ------- This branch splits all the analog signaling logic out of chan_dahdi.c into sig_analog.c. Functionality in theory should not change at all. Diffs (updated) ----- /trunk/channels/Makefile 196825 /trunk/channels/chan_dahdi.c 196825 /trunk/channels/sig_analog.h PRE-CREATION /trunk/channels/sig_analog.c PRE-CREATION Diff: http://reviewboard.digium.com/r/253/diff Testing ------- The main testing I did was simply making calls. Once I got that working most everything else seemed to work. Specifically made sure caller id works for both sending and receiving as well. Thanks, Jeff _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev

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