----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4466/#review14647 -----------------------------------------------------------
Ship it! Cool! I would suggest putting a note in UPGRADE.txt about the first call ID being 1 instead of 0. As minor as it is, it's a good idea for that to be documented in the upgrade notes. - Mark Michelson On March 11, 2015, 9:27 p.m., Corey Farrell wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/4466/ > ----------------------------------------------------------- > > (Updated March 11, 2015, 9:27 p.m.) > > > Review request for Asterisk Developers. > > > Bugs: ASTERISK-24833 > https://issues.asterisk.org/jira/browse/ASTERISK-24833 > > > Repository: Asterisk > > > Description > ------- > > The logger currently uses an AO2 to store a callid (int). This patch changes > callid's to 'unsigned int'. AO2 is not needed for callid's. > > This also fixes a theoretical infinate loop if a callid is subjected to an > extra unreference but left in threadstorage. If ast_log tries getting an > already released callid from threadstorage, ao2_ref will log an error. This > would cause ast_log to try grabbing a reference to the callid again (same > thread). The probability of this is very low and requires other code to be > broken, but it's not impossible. > > Change in behaviour: currently the first callid == 0. This patch changes the > first callid to 1 since 0 now represents the lack of callid. > > > Diffs > ----- > > /trunk/res/ari/resource_bridges.c 432661 > /trunk/main/pbx.c 432661 > /trunk/main/logger.c 432661 > /trunk/main/features.c 432661 > /trunk/main/dial.c 432661 > /trunk/main/core_unreal.c 432661 > /trunk/main/core_local.c 432661 > /trunk/main/cli.c 432661 > /trunk/main/channel_internal_api.c 432661 > /trunk/main/channel.c 432661 > /trunk/main/bridge_channel.c 432661 > /trunk/main/bridge_basic.c 432661 > /trunk/main/bridge.c 432661 > /trunk/main/autoservice.c 432661 > /trunk/include/asterisk/logger.h 432661 > /trunk/include/asterisk/core_unreal.h 432661 > /trunk/include/asterisk/channel.h 432661 > /trunk/include/asterisk/bridge_channel.h 432661 > /trunk/include/asterisk/bridge.h 432661 > /trunk/channels/sip/include/sip.h 432661 > /trunk/channels/sip/include/dialog.h 432661 > /trunk/channels/sip/dialplan_functions.c 432661 > /trunk/channels/sig_ss7.c 432661 > /trunk/channels/sig_pri.c 432661 > /trunk/channels/sig_analog.c 432661 > /trunk/channels/chan_sip.c 432661 > /trunk/channels/chan_motif.c 432661 > /trunk/channels/chan_iax2.c 432661 > /trunk/channels/chan_dahdi.c 432661 > /trunk/channels/chan_bridge_media.c 432661 > /trunk/apps/confbridge/conf_chan_announce.c 432661 > /trunk/apps/app_mixmonitor.c 432661 > > Diff: https://reviewboard.asterisk.org/r/4466/diff/ > > > Testing > ------- > > Ran a couple tests against chan_sip and chan_iax2, visually inspected log > files. > My dev system doesn't build chan_dahdi (ever), so Matt Jordan verified it > still compiles. > > > Thanks, > > Corey Farrell > >
-- _____________________________________________________________________ -- 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