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2010-09-16Merged revisions 287114 via svnmerge from mnicholson1-10/+10
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r287114 | mnicholson | 2010-09-16 14:52:39 -0500 (Thu, 16 Sep 2010) | 8 lines Don't stop printing cdr variables if we encounter one with a blank name or value. (closes issue #17900) Reported by: under Patches: core-show-channel-cdr-fix1.diff uploaded by mnicholson (license 96) Tested by: mnicholson ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@287115 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-15Merged revisions 286941 via svnmerge from jpeeler1-5/+29
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286941 | jpeeler | 2010-09-15 15:08:52 -0500 (Wed, 15 Sep 2010) | 7 lines Ensure mailbox is not filled to capacity before doing message forwarding. Specifically, before prompting to record a prepended message the capacity is checked first. If the mailbox is full the extension will be reprompted. ABE-2517 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@286998 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-14Merged revisions 286756 via svnmerge from mnicholson1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286756 | mnicholson | 2010-09-14 14:26:18 -0500 (Tue, 14 Sep 2010) | 13 lines Don't clear the username from a realtime database when a registration expires. Non-realtime chan_sip does not clear the username from memory when a registration expiries so realtime probably shouldn't either. (closes issue #17551) Reported by: ricardolandim Patches: reg-expiry-username-1.4-fix1.diff uploaded by mnicholson (license 96) reg-expiry-username-1.6.2-fix1.diff uploaded by mnicholson (license 96) reg-expiry-username-1.8-fix1.diff uploaded by mnicholson (license 96) reg-expiry-username-trunk-fix1.diff uploaded by mnicholson (license 96) Tested by: ricardolandim, mnicholson ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@286757 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-14Merged revisions 286679 via svnmerge from mnicholson1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286679 | mnicholson | 2010-09-14 13:00:01 -0500 (Tue, 14 Sep 2010) | 7 lines Only drop duplicate answer frames if the channel is bridged. Back in r3710 ast_read() was modified to drop answer frames on channels that were in the UP state. This modification prevented bridges that were up before the answer from being broken and reestablished by an ANSWER control frame. That change also prevents pickup of channels called from the ast_dial framework from working properly. The ast_dial framework expects to see an ANSWER frame after dialing and the pickup code queues one but ast_read() drops it. This new change only drops ANSWER frames when the channel is bridged, allowing the answer queued by the pickup code to properly pass through ast_read() on to the ast_dial framework. ABE-2473 (related to issue #2342) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@286681 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-14Add documentation on missing backend tables for Voicemailtilghman2-0/+58
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@286587 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-13C precedence got metilghman1-3/+3
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@286557 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-13Refactor conversion to ast_poll() to fix callparking regression.tilghman1-21/+23
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@286527 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-13Remove "Internal IP" from sip show settings, as it's not at all useful to ↵qwell1-1/+0
display. (closes issue #17840) Reported by: oej git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@286456 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-13Blocked revisions 286381 via svnmergeqwell0-0/+0
........ r286381 | qwell | 2010-09-13 10:12:51 -0500 (Mon, 13 Sep 2010) | 5 lines Add stuff to svn:ignore for tests/ directory. (closes issue #17983) Reported by: oej ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@286382 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-11Merged revisions 286267 via svnmerge from oej1-1/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286267 | oej | 2010-09-11 18:59:20 +0200 (Lör, 11 Sep 2010) | 4 lines Handle error response when we can't make file compatible Review: https://reviewboard.asterisk.org/r/911/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@286268 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-10Merged revisions 286222 via svnmerge from twilson1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286222 | twilson | 2010-09-10 17:54:23 -0500 (Fri, 10 Sep 2010) | 1 line Return -1 if chan_local doesn't support an option ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@286223 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-10Merged revisions 286114 via svnmerge from pabelanger1-19/+21
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286114 | pabelanger | 2010-09-10 16:35:08 -0400 (Fri, 10 Sep 2010) | 4 lines Load iax.conf before registering any functions/applications/actions. Review: https://reviewboard.asterisk.org/r/914/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@286117 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-10Merged revisions 286113 via svnmerge from rmudgett1-0/+16
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286113 | rmudgett | 2010-09-10 15:33:16 -0500 (Fri, 10 Sep 2010) | 11 lines An outgoing call may not get hung up if a pre-connect incoming ISDN call is disconnected. If the ISDN link a pre-connect incoming call is using fails or is reset, the outgoing leg may not hang up or be delayed in hanging up. (Causes: PRI_CAUSE_NETWORK_OUT_OF_ORDER, PRI_CAUSE_DESTINATION_OUT_OF_ORDER, and PRI_CAUSE_NORMAL_TEMPORARY_FAILURE.) Just hang up the call if the incoming call leg hangs up before connecting for any reason. It makes no sense to send a BUSY or CONGESTION control frame to the outgoing call leg under these circumstances. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@286116 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-10Merged revisions 286059 via svnmerge from twilson5-3/+113
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286059 | twilson | 2010-09-10 14:25:08 -0500 (Fri, 10 Sep 2010) | 16 lines Inherit CHANNEL() writes to both sides of a Local channel Having Local (/n) channels as queue members and setting the language in the extension with Set(CHANNEL(language)=fr) sets the language on the Local/...,2 channel. Hold time report playbacks happen on the Local/...,1 channel and therefor do not play in the specified language. This patch modifies func_channel_write to call the setoption callback and pass the CHANNEL() write info to the callback. chan_local uses this information to look up the other side of the channel and apply the same changes to it. (closes issue #17673) Reported by: Guggemand Review: https://reviewboard.asterisk.org/r/903/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@286115 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-10Blocked revisions 286070 via svnmergedvossel0-0/+0
........ r286070 | dvossel | 2010-09-10 15:03:50 -0500 (Fri, 10 Sep 2010) | 32 lines Fixes sip extension state update DEADLOCK PROBLEM: In chan_sip, and all the other channel drivers, it is common for us to hold the tech_pvt lock while we ask the Asterisk core about an extension and context. Every time we do this the locking order becomes, (1. tech_pvt lock ---> 2. global context lock). In chan_sip when a dialog subscribes to a hint, that locking order is reversed in the extensionstate callback which will occur outside of the channel_driver's monitor loop. So, on an extension state update we have (1. global context lock ----> 2. tech_pvt lock). Typically when we have to do a reversed locking order like this we'd just do some sort of deadlock avoidance to fix the problem... That will not work here. There are more locks involved here than just the context and tech_pvt. Those are the two that are colliding, but it is impossible to give up the context lock because the global hints list lock MUST be held as well and we can not give that lock up during the extensionstate callback traversal... The locking order for the context and hints are (1. global context lock ----> 2. hints list lock). Deadlock avoidance is not an option here. SOLUTION: The solution this patch implements is to queue the extension state updates into a list and send the NOTIFY messages out during the do_monitor pvt traversal. This clears out the problem of having to hold the context lock before the tech_pvt lock entirely. (closes issue #17888) Reported by: zerohalo ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@286071 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-10Merged revisions 286023 via svnmerge from tilghman2-4/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286023 | tilghman | 2010-09-10 13:22:04 -0500 (Fri, 10 Sep 2010) | 2 lines Missing newline ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@286024 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-10Another fix for Mac OS X.tilghman1-6/+4
While trying to fix this the "right" way, I wandered into dependency hell. Two hours later, I backed out, and just removed the offending code. ast_inline_api only goes one level deep and then it breaks. Ouch. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@285961 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-10Merged revisions 285889 via svnmerge from tilghman5-11/+1591
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285889 | tilghman | 2010-09-09 19:13:45 -0500 (Thu, 09 Sep 2010) | 7 lines Fix Mac OS X build. This also fixes a rather grievous calculation error for the offset of ast_fdset, which was masked on Linux and FreeBSD, because these platforms check the first 256 FDs regardless of the bitmask setting (due to backwards compatibility). ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@285930 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-09Merged revisions 285817 via svnmerge from pabelanger1-0/+8
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285817 | pabelanger | 2010-09-09 18:34:35 -0400 (Thu, 09 Sep 2010) | 8 lines GCC 4.2.x optimizations result in improper behavior of GSM codec (closes issue #17688) Reported by: pprindeville Patches: asterisk-trunk-bugid11243.patch uploaded by pprindeville (license 347) Tested by: mkeuter, pprindeville ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@285818 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-09Merged revisions 285742 via svnmerge from qwell1-1/+13
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285742 | qwell | 2010-09-09 15:06:31 -0500 (Thu, 09 Sep 2010) | 9 lines Transmit silence when reading DTMF in ast_readstring. Otherwise, you could get issues with DTMF timeouts causing hangups. (closes issue #17370) Reported by: makoto Patches: channel-readstring-silence-generator.patch uploaded by makoto (license 38) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@285744 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-09Fixes an issue with dialplan pattern matching where the specificity for ↵bbryant48-7445/+27059
pattern ranges and pattern special characters was inconsistent. (closes issue #16903) Reported by: Nick_Lewis Patches: pbx.c-specificity.patch uploaded by Nick Lewis (license 657) Tested by: Nick_Lewis git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@285710 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-03Merged revisions 280811 via svnmerge from tilghman2-3/+9
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r280811 | tilghman | 2010-08-03 15:49:10 -0500 (Tue, 03 Aug 2010) | 9 lines Prevent DAHDI channels from overriding the callerid, once it's been set by the user. (closes issue #16661) Reported by: jstapleton Patches: 20100414__issue16661.diff.txt uploaded by tilghman (license 14) 20100415__issue16661__1.6.2.diff.txt uploaded by tilghman (license 14) Tested by: jstapleton ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@280812 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-03Document -B and -W flags and regenerate manpage from sgmltilghman3-119/+204
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@280739 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-02Allow the pipe, but also allow the commatilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@280671 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-02Change SIP NOTIFY requests to expect a response so authentication will work.jpeeler1-1/+1
This changes the request to be sent with the transmit type XMIT_RELIABLE so that sip_ack doesn't return false and cause the 401 to be ignored in cases where authentication is required. (closes issue #14255) Reported by: zktech git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@280669 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-29Off-by-one errortilghman1-8/+8
(closes issue #17590) Reported by: atis Patches: 20100729__issue17590.diff.txt uploaded by tilghman (license 14) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@280556 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-29fixes wrong SRV query for TLS connectiondvossel1-5/+33
(closes issue #17612) Reported by: marcelloceschia Patches: chan-sip_srvQuery.patch uploaded by marcelloceschia (license 1079) chan-sip_Trunk_srvQuery.patch uploaded by st (license 907) chan-sip_asterisk18b1_srvQuery.patch uploaded by marcelloceschia (license 1079) Tested by: marcelloceschia, st, pabelanger git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@280551 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-29Merged revisions 280448 via svnmerge from dvossel1-0/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r280448 | dvossel | 2010-07-29 14:04:23 -0500 (Thu, 29 Jul 2010) | 12 lines fixes issue with translator frame not getting freed A translator frame even if it local storage so the translation path can be freed. This issue prevented g729 licenses from being freed up. (closes issue #17630) Reported by: manvirr Patches: encoder_fix.diff uploaded by dvossel (license 671) Tested by: manvirr, dvossel ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@280449 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-29Merged revisions 280341 via svnmerge from jeang1-0/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r280341 | jeang | 2010-07-29 10:52:31 -0500 (Thu, 29 Jul 2010) | 2 lines Fix a dsp structure leak occuring when a local channel is put into a meetme conference, then masquaraded away. ABE-2422 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@280345 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-29Implement support for ast_channel_queryoption on local channels. Currently ↵mnicholson1-0/+54
only AST_OPTION_T38_STATE is supported. ABE-2229 Review: https://reviewboard.asterisk.org/r/813/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@280306 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-28Work around some silly behavior on BSD.qwell1-2/+2
A non-zero exit from a subshell should make the build fail. (closes issue #17621) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@280231 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-28Add missing enum value "unknown" to the SS7 called_nai and calling_nai ↵rmudgett1-0/+4
config options. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@280229 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-28Add sha1sum-sh in case there is no util on the system.qwell1-0/+71
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@280227 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-28Remove unnecessary subshells. Attempt to make checksumming work.qwell1-28/+31
Also improves readability. (issue #17621) Reported by: bjm Review: https://reviewboard.asterisk.org/r/808/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@280193 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-28Plug a reference leak in app_queue when adding members dynamically.seanbright1-0/+1
(closes issue #17738) Reported by: bobwienholt Patches: issue17738.patch uploaded by bobwienholt (license 950) Tested by: bobwienholt, seanbright git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@280160 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-28Merged revisions 280088 via svnmerge from lmadsen1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r280088 | lmadsen | 2010-07-28 08:50:38 -0500 (Wed, 28 Jul 2010) | 1 line Update help text to be less confusing. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@280089 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-27Merged revisions 279945 via svnmerge from dvossel3-1/+28
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r279945 | dvossel | 2010-07-27 15:33:40 -0500 (Tue, 27 Jul 2010) | 19 lines remove empty audiohook write list on channel If a channel has an audiohook write list created on it, that list stays on the channel until the channel is destroyed. There is no reason to keep that list on the channel if it becomes empty. If it is empty that just means we are doing needless translating for every ast_read and ast_write. This patch removes the audiohook list from the channel once it is detected to be empty on either a read or write. If a audiohook is added back to the channel after this list is destroyed, the list just gets recreated as if it never existed to begin with. (closes issue #17630) Reported by: manvirr Review: https://reviewboard.asterisk.org/r/799/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@279946 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-27Add SHA1SUM to configure, since we require it for sounds/qwell3-1/+105
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@279883 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-27Remove aptly-named EMPTY and BS vars, since they aren't used anymore.qwell1-3/+0
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@279852 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-27Simply sounds/Makefile some more.qwell1-16/+16
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@279849 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-27Fix bad behavior of dynamic_exclude_static option in sip.conf.mmichelson1-7/+7
We were attempting to create a contactdeny rule based on the peer's IP address before the peer's IP address had been set. By moving the processing further down in the function, we can ensure stuff works as we expect for it to. (closes issue #17717) Reported by: mmichelson Patches: 17717.patch uploaded by mmichelson (license 60) Tested by: DennisD git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@279784 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-26Really fix sounds Makefile (and make it readableish).qwell5-25509/+6749
There was a rather large syntax error that should have caused ALL versions of GNU make to fail. I don't know how it worked. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@279657 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-26Dunno why this worked on my machine, but it works better this way.tilghman2-5/+5
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@279609 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-26Apply all patches in:ghenry1-1/+18
https://issues.asterisk.org/view.php?id=13573 (closes issue #13573) Reported by: navkumar Patches: res_config_ldap-category.diff uploaded by navkumar (license 580) res_config_ldap.patch uploaded by bencer (license 961) res_config_ldap uploaded by bencer (license 961) Tested by: suretec git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@279597 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-26Use a special Makefile for noobs who still have GNU Make 3.80.tilghman4-2/+180
(Closes issue #17716) Reported by: farisraouf git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@279561 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-26Expand the correct value within AST_OPTION_ONLY.seanbright1-1/+1
(closes issue #17703) Reported by: stuarth git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@279501 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-24Minor update to man pagesnuffy1-3/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@279347 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-24Blocked revisions 279344 via svnmergejpeeler0-0/+0
........ r279344 | jpeeler | 2010-07-24 18:27:22 -0500 (Sat, 24 Jul 2010) | 4 lines Provide a default value for DAHDI_TRANSCODE so when DAHDI is not installed menuselect doesn't get confused: Unknown value '' found in build_tools/menuselect-deps for DAHDI_TRANSCODE ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@279345 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-23Merged revisions 279206 via svnmerge from rmudgett2-6/+12
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r279206 | rmudgett | 2010-07-23 16:56:44 -0500 (Fri, 23 Jul 2010) | 7 lines SIP promiscuous redirect could fail to dial the redirect. The ast_channel was created with one variable to ast_request() but the call to ast_call() that initiates the outgoing call was using a different variable. The two variables are not equivalent if the call_forward string included a channel technology specifier. e.g., SIP/200 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@279207 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-23Backport sip_uri_params_cmp() fix from trunk to 1.6.2.mmichelson1-39/+25
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@279112 f38db490-d61c-443f-a65b-d21fe96a405b
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