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2011-06-17Merge 192.168.0.100:/repos/git/asteriskPatrick McHardy30-411/+1101
2011-06-16Merged revisions 324048 via svnmerge from twilson4-25/+45
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324048 | twilson | 2011-06-16 17:35:41 -0500 (Thu, 16 Jun 2011) | 8 lines Lock the channel before calling the setoption callback The channel needs to be locked before calling these callback functions. Also, sip_setoption needs to lock the pvt and a check p->rtp is non-null before using it. Review: https://reviewboard.asterisk.org/r/1220/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@324050 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-16Merged revisions 323990 via svnmerge from rmudgett1-0/+3
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323990 | rmudgett | 2011-06-16 13:12:32 -0500 (Thu, 16 Jun 2011) | 5 lines The test_event unit test is occasionally failing. Wait for the special posted event to process before adding a new subscription. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323991 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-16Merged revisions 323932 via svnmerge from twilson1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323932 | twilson | 2011-06-16 10:58:22 -0500 (Thu, 16 Jun 2011) | 4 lines Don't assume ASTDBDIR exists It most likely doesn't on FreeBSD ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323933 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-15Merged revisions 323866 via svnmerge from twilson1-4/+4
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323866 | twilson | 2011-06-15 15:03:58 -0500 (Wed, 15 Jun 2011) | 2 lines Remove now-useless cast of ARRAY_LEN ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323867 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-15Merged revisions 323863 via svnmerge from twilson1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323863 | twilson | 2011-06-15 14:58:18 -0500 (Wed, 15 Jun 2011) | 2 lines Make ARRAY_LEN() return the same type on x86 and x86_64 systems ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323864 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-15Merged revisions 323859 via svnmerge from twilson1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323859 | twilson | 2011-06-15 14:45:20 -0500 (Wed, 15 Jun 2011) | 2 lines Fix more ARRAY_LEN format string issues ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323860 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-15Merged revisions 323754 via svnmerge from twilson1-1/+5
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r323754 | twilson | 2011-06-15 13:21:52 -0500 (Wed, 15 Jun 2011) | 23 lines Merged revisions 323733 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r323733 | twilson | 2011-06-15 13:13:00 -0500 (Wed, 15 Jun 2011) | 16 lines Merged revisions 323732 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r323732 | twilson | 2011-06-15 13:06:24 -0500 (Wed, 15 Jun 2011) | 9 lines Fix DYNAMIC_FEATURES DYNAMIC_FEATURES were broken by a recent DTMF change. This patch makes sure that dynamic features are also checked when deciding whether or not to pass DTMF through or store it for interpreting. (closes issue ASTERISK-17914) Reported by: vrban ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323760 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-15Blocked revisions 323730 via svnmergejrose0-0/+0
........ r323730 | jrose | 2011-06-15 12:42:42 -0500 (Wed, 15 Jun 2011) | 11 lines Adds locking to find_table in res_configure_pgsql to prevent a crash. Bryonclark described the problem as occuring during this function because of multiple simultaneous database operations causing corruption against a pgsqlConn object. (closes issue ASTERISK-17811) Reported by: byronclark Patches: pgsql_find_table_locking.patch uploaded by byronclark (license 1200) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323731 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-15Merged revisions 323672 via svnmerge from twilson1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323672 | twilson | 2011-06-15 10:09:51 -0700 (Wed, 15 Jun 2011) | 5 lines Cast ARRAY_LEN to size_t for ast_logging 32-bit and 64-bit machines return different types for ARRAY_LEN(), so cast it before using in a format string. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323673 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-15Merged revisions 323669-323670 via svnmerge from rmudgett2-30/+81
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323669 | rmudgett | 2011-06-15 11:43:18 -0500 (Wed, 15 Jun 2011) | 21 lines [regression] Voicemail MWI is no longer sent. When leaving a voicemail, the MWI message is never sent. The same thing happens when checking a voicemail and marking it as read. If you restart Asterisk, everything comes up at that state correctly, but changes to the messages in voicemail causes the light to not be set appropriately. Very easy to reproduce. * Made ast_event_check_subscriber() return TRUE if there are ANY subscribers to an event type when there are no restricting ie values passed. This allows an event being queued to be queued. (closes issue ASTERISK-18002) Reported by: lmadsen Tested by: lmadsen, irroot Patches: jira_asterisk_18002_v1.8.patch uploaded by rmudgett (License #5621) (closes issue ASTERISK-18019) ........ r323670 | rmudgett | 2011-06-15 11:43:31 -0500 (Wed, 15 Jun 2011) | 7 lines Add a test to the event unit tests to catch ASTERISK-18002. The new tests check to see if there are ANY subscribers to the event type when ast_event_check_subscriber() is not passed any specific ie values. (issue ASTERISK-18002) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323671 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-15Merged revisions 323610 via svnmerge from jrose1-6/+38
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323610 | jrose | 2011-06-15 11:09:24 -0500 (Wed, 15 Jun 2011) | 7 lines Adds PQclear calls on result to various parts of res_conf_pgsql (closes issue ASTERISK-17812) Reported by: byronclark Patches: pgsql_pqclear.patch uploaded by byronclark (license 1200) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323621 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-15Merged revisions 323608 via svnmerge from seanbright1-43/+35
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r323608 | seanbright | 2011-06-15 11:31:53 -0400 (Wed, 15 Jun 2011) | 39 lines Merged revisions 323579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r323579 | seanbright | 2011-06-15 11:22:50 -0400 (Wed, 15 Jun 2011) | 32 lines Merged revisions 323559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r323559 | seanbright | 2011-06-15 11:15:30 -0400 (Wed, 15 Jun 2011) | 25 lines Resolve a segfault/bus error when we try to map memory that falls on a page boundary. The fix for ASTERISK-15359 was incorrect in that it added 1 to the length of the mmap'd region. The problem with this is that reading/writing to that extra byte outside of the bounds of the underlying fd causes a bus error. The real issue is that we are working with both a FILE * and the raw fd underneath it and not synchronizing between them. The code that was removed in ASTERISK-15359 was correct, but we weren't flushing the FILE * before mapping the fd. Looking at the manager code in 1.4 reveals that the FILE * in 'struct mansession' is never used except to create a temporary file that we immediately fdopen. This means we just need to write a 0 byte to the fd and everything will just work. The other branches require a call to fflush() which, while not a guaranteed fix, should reduce the likelihood of a crash. This all makes sense in my head. (closes issue ASTERISK-16460) Reported by: Ravelomanantsoa Hoby (hoby) Patches: issue17747_1.4_svn_markII.patch uploaded by Sean Bright (license #5060) ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323609 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-15CONFBRIDGE_INFO function to get conference datakmoore2-0/+90
Added the CONFBRIDGE_INFO dialplan function to get information about a conference bridge including locked status and number of parties, admins, and marked users. Review: https://reviewboard.asterisk.org/r/1271/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323517 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-15Merged revisions 323456 via svnmerge from rmudgett1-2/+3
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323456 | rmudgett | 2011-06-14 19:50:20 -0500 (Tue, 14 Jun 2011) | 1 line Add missing break in ast_event_get_cached(). ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323457 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-14Merged revisions 323392,323394 via svnmerge from rmudgett2-7/+14
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323392 | rmudgett | 2011-06-14 12:21:24 -0500 (Tue, 14 Jun 2011) | 6 lines Add more strict hostname checking to ast_dnsmgr_lookup(). Change suggested in review. Review: https://reviewboard.asterisk.org/r/1240/ ........ r323394 | rmudgett | 2011-06-14 12:21:39 -0500 (Tue, 14 Jun 2011) | 2 lines Made ast_sockaddr_split_hostport() port warning msgs more meaningful. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323397 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-14Merged revisions 323370 via svnmerge from twilson4-1/+131
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323370 | twilson | 2011-06-14 09:33:55 -0700 (Tue, 14 Jun 2011) | 10 lines Add rtpkeepalives back to 1.8 The RTP-engine conversion left out support for handling rtpkeepalives. This patch adds them back. (closes issue ASTERISK-17304) Reported by: lmadsen Review: https://reviewboard.asterisk.org/r/1226/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323374 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-14Merged revisions 323371 via svnmerge from jrose1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323371 | jrose | 2011-06-14 11:38:43 -0500 (Tue, 14 Jun 2011) | 12 lines Changes contact use in build_peer to use the FORCE_RPORT flag instead of RPORT_PRESENT It turned out that this was causing NAT=Yes to always use rport when present which was against 1.6.2 behavior and the check itself was redundant since the only way this segment of code could be reached was if RPORT_PRESENT was already evaluated as true earlier. (closes issue ASTERISK-17789) Reported by: byronclark Patches: use_sip_nat_force_rport.patch uploaded by byronclark (license 1200) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323372 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-14Store sip peer name as var data on a outofcall msg.dvossel1-0/+5
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323325 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-13Config inheritance doesn't work with ConfBridge() menu definitionskmoore1-1/+11
Current behavior in ConfBridge menu definitions is that first definition takes precedence, even in templated situations. This change allows inheritance and overriding to work as expected so that the last definition takes precedence. (closes ASTERISK-17986) Review: https://reviewboard.asterisk.org/r/1267/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323272 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-13Merged revisions 323213 via svnmerge from lmadsen1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323213 | lmadsen | 2011-06-13 15:51:52 -0400 (Mon, 13 Jun 2011) | 6 lines Avoid dividing by zero with L() option to Dial() Reported by: nicolasom Patches: issue-17995.patch - nicolasom (License #5994) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323214 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-13Addition of "outofcall_message_context" sip.conf option.dvossel4-4/+33
Review: https://reviewboard.asterisk.org/r/1265/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323212 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-13Merged revisions 323154 via svnmerge from lmadsen1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323154 | lmadsen | 2011-06-13 15:00:41 -0400 (Mon, 13 Jun 2011) | 6 lines Tweak documentation for AGI Hangup command. (closes issue ASTERISK-17999) Reported by: Ben Klang Patches: hangup-doc.diff - uploaded by Ben Klang (License #5876) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323155 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-13MOH for only user not working with ConfBridgekmoore2-4/+17
This adds the playing_moh flag to the conference_bridge_user struct that signifies when MOH should be playing so code doesn't have to guess whether MOH is playing. This change also adds the necessary checking to ensure that MOH continues playing for a single user in a conference after the join sound is played when configured to do so. (closes ASTERISK-17988) Review: https://reviewboard.asterisk.org/r/1263/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323107 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-13ConfBridge: Use of bridge or user profiles that don't existkmoore1-2/+10
Bridge and user profiles are not checked for existence before use. The lack of a fully formed bridge profile can cause a segfault when sounds are accessed. This change ensures that bridge and user profiles exist prior to usage attempts. Review: https://reviewboard.asterisk.org/r/1264/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323106 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-10Merged revisions 323040 via svnmerge from mnicholson1-3/+10
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323040 | mnicholson | 2011-06-10 14:20:41 -0500 (Fri, 10 Jun 2011) | 5 lines Unlock the sip channel during fax detection like chan_dahdi does to prevent a deadlock with ast_autoservice_stop. (closes issue ASTERISK-17798) tested by mnicholson ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@323041 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-10Merged revisions 322981 via svnmerge from twilson1-145/+154
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322981 | twilson | 2011-06-10 08:29:00 -0700 (Fri, 10 Jun 2011) | 11 lines Avoid a DB1 infinite loop bug Explicity check the last entry in the DB and make sure that we don't iterate past it. Since there can be no duplicates, this just makes sure that we stop after matching the last key. This patch also refactors the code to get away from some code duplication. A previous patch added many astdb tests and this patch passed them. Review: https://reviewboard.asterisk.org/r/1259/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322982 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-10Merged revisions 322923 via svnmerge from twilson1-0/+222
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322923 | twilson | 2011-06-09 19:33:23 -0700 (Thu, 09 Jun 2011) | 2 lines Add some astdb unit tests ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322940 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-09Merged revisions 322865 via svnmerge from twilson1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322865 | twilson | 2011-06-09 15:29:20 -0700 (Thu, 09 Jun 2011) | 4 lines Correct ast_db_deltree documentation ast_db_deltree returns -1 on error, otherwise the number of deletions ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322866 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-09Merged revisions 322807 via svnmerge from mnicholson1-6/+7
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322807 | mnicholson | 2011-06-09 12:37:07 -0500 (Thu, 09 Jun 2011) | 5 lines don't drop any voice frames when checking for T.38 during early media (closes issue ASTERISK-17705) Review: https://reviewboard.asterisk.org/r/1186/ patch by oej reported by oej ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322808 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-09Merged revisions 322749 via svnmerge from rmudgett3-45/+92
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322749 | rmudgett | 2011-06-09 11:31:53 -0500 (Thu, 09 Jun 2011) | 15 lines Remove potential deadlock in call pickup race. Deadlock is possible in ast_do_pickup() when holding the target channel lock and trying to get the chan channel lock. Also, holding the target lock when calling ast_channel_masquerade() is not a good idea because that routine does deadlock avoidance. * Removed the need to hold the target lock after marking the target with a datastore and getting the connected line data off of the target channel. * Moved can_pickup() to ast_can_pickup() in features.c. Now all the call pickup methods use the same basic call pickup availability check. Review: https://reviewboard.asterisk.org/r/1234/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322750 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-09Blocked revisions 322585 via svnmergejrose0-0/+0
........ r322585 | jrose | 2011-06-09 09:06:42 -0500 (Thu, 09 Jun 2011) | 11 lines Adds ast_escape_encoded utility to properly handle escaping of quoted field before uri. This commit backports a feature in trunk affecting initreqprep so that display name won't be encoded improperly. Also includes unit tests for the ast_escape_quoted function. This patch gives 1.8 a much improved outlook in countries which don't use standard ASCII characters. (closes issue ASTERISK-16949) Reported by: Örn Arnarson Review: https://reviewboard.asterisk.org/r/1235/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322586 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-09Add autoanswer to skinny.wedhorn1-5/+74
Autoanswer added to skinny based on incoming chan var SKINNY_AUTOANSWER. Initial value must be the time to autoanswer in ms, then optionally :BEEP to play a tone when answered and :MUTE to mute the mic when answering. eg 3000:MUTE:BEEP will ring for 3 secs, then answer, mute the mic, and play a beep. just 3000 would answer afer 3 secs of ringing with no beep and full two way audio. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322544 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-08Merged revisions 322484 via svnmerge from rmudgett1-1/+3
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322484 | rmudgett | 2011-06-08 15:46:55 -0500 (Wed, 08 Jun 2011) | 15 lines Ring all queue with more than 255 agents will cause crash. 1. Create a ring-all queue with 500 permanent agents. 2. Call it. 3. Asterisk will crash. The watchers array in app_queue.c has a hard limit of 255. Bounds checking is not done on this array. No sane person should put 255 people in a ring-all queue, but we should not crash anyway. * Added bounds checking to the watchers array. JIRA AST-464 JIRA SWP-2903 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322485 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-08Merged revisions 322425 via svnmerge from rmudgett1-0/+8
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322425 | rmudgett | 2011-06-08 13:46:30 -0500 (Wed, 08 Jun 2011) | 16 lines SRV lookup attempted for SIP peers listed as an IP address. Asterisk attempts to SRV lookup a host name even if the host name is an IP address. Regression introduced when IPv6 support was added. * Restored the check in ast_dnsmgr_lookup() to see if the given host name is an IP address. The IP address could be in either IPv4 or IPv6 formats. (closes issue ASTERISK-17815) Reported by: Byron Clark Tested by: Byron Clark, Richard Mudgett Patches: issue19248_v1.8.patch - uploaded by Richard Mudgett (License #5621) Review: https://reviewboard.asterisk.org/r/1240/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322426 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-08Merge 192.168.0.100:/repos/git/asteriskPatrick McHardy182-4597/+14845
2011-06-08Remove skinny do_monitor and use ast_sched_start insteadwedhorn1-76/+5
The do_monitor seemed to be there for task scheduling and network monitoring. However, the network monitoring has a dedicated thread so the ast_io_wait was basically just a usleep as it didn't actually seem to be monitoring anything. Review: https://reviewboard.asterisk.org/r/1256/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322381 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-08Merged revisions 322322 via svnmerge from irroot1-2/+17
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322322 | irroot | 2011-06-08 08:18:38 +0200 (Wed, 08 Jun 2011) | 18 lines Make handle_request_publish do dialog expiration and destruction. This patch fixes handle_request_publish so that it does dialog expiration and destruction. Without this patch the incoming PUBLISH requests will get stuck in the dialog list. Restarting asterisk is the only way to remove them. Personal observation on one system the server hung up while looping through the channels rendering asterisk unusable and all sip phones unregisterd when they try reregister more requests are added. (closes issue #18898) Reported by: gareth Tested by: loloski, Chainsaw, wimpy, se, kuj, irroot Jira: https://issues.asterisk.org/jira/browse/ASTERISK-17915 Review: https://reviewboard.asterisk.org/r/1253 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322323 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-07Correct some whitespace and a reference debug message.rmudgett2-14/+14
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322284 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-07Actually check the "sendtodialplan" option setting for xmpp.russell1-12/+14
(closes issue ASTERISK-17978) Reported by: elguero Patches: stop_messages_going_to_dialplan.patch (license #5026) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322244 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-07Merged revisions 322189 via svnmerge from pabelanger1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322189 | pabelanger | 2011-06-07 13:59:13 -0400 (Tue, 07 Jun 2011) | 4 lines Use correct syntax for 'sip notify snom-reboot' (closes issue ASTERISK-17915) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322190 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-06Remove Unused Var Warning rt_handle_member_recordirroot1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322128 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-06Refactor rt_handle_member_recordirroot1-14/+11
Review: https://reviewboard.asterisk.org/r/1172 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322111 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-06Merged revisions 322069 via svnmerge from jrose2-2/+12
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322069 | jrose | 2011-06-06 14:07:56 -0500 (Mon, 06 Jun 2011) | 8 lines Fixes level toggling for logger set levels since it was reversed (closes issue ASTERISK-17850) Reported by: Luke H Tested by: jrose, Luke H Review: https://reviewboard.asterisk.org/r/1244/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322070 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-03Merged revisions 321926 via svnmerge from rmudgett2-7/+43
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321926 | rmudgett | 2011-06-03 17:09:36 -0500 (Fri, 03 Jun 2011) | 18 lines Asterisk crash when unloading cdr_radius/cel_radius. The rc_openlog() API call is passed a string that is used by openlog() to format log messages. The openlog() does not copy the string it just keeps a pointer to it. When the module is unloaded, the string is gone from memory. Depending upon module load order and if the other module then has an error, a crash happens. * Pass rc_openlog() a strdup'd string with the understanding that there will be a small memory leak if the cdr_radius/cel_radius modules are unloaded. * Call rc_destroy() to free the rc handle memory when the module is unloaded. JIRA AST-483 JIRA SWP-3062 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@321927 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-03Merged revisions 321924 via svnmerge from rmudgett1-7/+10
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321924 | rmudgett | 2011-06-03 16:49:17 -0500 (Fri, 03 Jun 2011) | 5 lines Be more explicit for CCSS generic device state event subscription. Make CCSS generic device state event subscription specify the AST_EVENT_IE_STATE ie exists to be safe. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@321925 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-03Merged revisions 321871 via svnmerge from rmudgett2-167/+594
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321871 | rmudgett | 2011-06-03 15:58:13 -0500 (Fri, 03 Jun 2011) | 27 lines Event subscription fixes. Must commit the subscription fixes together with the integration subscription tests. The subscription fixes cause an erroneously passing test to fail. The new subscription tests detect errors without the subscription fixes. * Added missing event_names[] table entry. * Reworked ast_event_check_subscriber()/match_sub_ie_val_to_event() to correctly detect if a subscriber exists for the proposed event. * Made match_ie_val() and match_sub_ie_val_to_event() check the buffer length for RAW payload types. * Fixed error handling memory leak in ast_event_sub_activate(), ast_event_unsubscribe(), and ast_event_queue(). * Made ast_event_new() and ast_event_check_subscriber() better protect themselves from an invalid payload type. * Added container lock protection between removing old cache events and adding the new cached event in ast_event_queue_and_cache()/event_update_cache(). * Added new event subscription tests. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@321872 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-03Merged revisions 321812-321813 via svnmerge from rmudgett4-11/+5
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321812 | rmudgett | 2011-06-03 14:55:21 -0500 (Fri, 03 Jun 2011) | 1 line Correct IAX2 and SIP event subscription description string. ........ r321813 | rmudgett | 2011-06-03 14:56:09 -0500 (Fri, 03 Jun 2011) | 1 line Constify subscription description parameter string. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@321814 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-03Blocked revisions 321753 via svnmergerussell0-0/+0
........ r321753 | russell | 2011-06-03 13:32:45 -0500 (Fri, 03 Jun 2011) | 2 lines Backport an astobj2 unit test so that it runs on 1.8 as well. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@321754 f38db490-d61c-443f-a65b-d21fe96a405b
2011-06-03Fix some astobj2 iterator breakage, add another unit test.russell2-5/+133
Review: https://reviewboard.asterisk.org/r/1254/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@321752 f38db490-d61c-443f-a65b-d21fe96a405b
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