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${ noResults }
18147 Commits (e1c03cbf1a699ce759337283e52e5e0e2ef9ac18)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
c524f905a6 |
Two fixes found while debugging with ast_backtrace():
1) If MALLOC_DEBUG is used when concurrently using ast_backtrace, the free() used in that routine will trigger an error, because the memory was allocated internally to libc, where we could not intercept that call to wrap it. Therefore, it's not memory we knew about, and the free is reported as an error. 2) Now that channels are objects, the old hack of initializing a channel to all zeroes no longer works, since we may try to call something like ast_channel_lock() within a function on that reference. In that case, it's reported as an error, because the pointer isn't an object reference. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194101 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
68a37433bc |
Fix a crash when logging out from the AMI and avoid astobj2 warning messages.
When the user logout the session was being destroyed twice and the file descriptor was being closed twice. The sessions reference counter wasn't used in a proper way. The 'mansession' structure was being treated as an astobj2 and we were calling ao2_lock/ao2_unlock causing astobj2 report a warning message and not locking the structure. Also we were using an ugly naming convention 'destroy_session', 'session_destroy', 'free_session', ... all this "duplicated" code was merged. (closes issue #14974) Reported by: pj Patches: manager.diff2 uploaded by eliel (license 64) Tested by: dhubbard, eliel, mnicholson (closes issue #15088) Reported by: eliel Review: http://reviewboard.asterisk.org/r/248/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194060 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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69976640f5 |
Merged revisions 194028 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194028 | mnicholson | 2009-05-12 17:15:45 -0500 (Tue, 12 May 2009) | 16 lines This change modifies app_queue to properly generate CDR records in failure situations. This involves setting a proper cdr disposition coresponding to the given failure condition and ensuring the proper information is stored in the cdr record. (closes issue #13691) Reported by: dferrer Tested by: mnicholson (closes issue #13637) Reported by: atis Tested by: atis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194057 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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84a4d2484d |
Merged revisions 193955 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193955 | tilghman | 2009-05-12 15:39:21 -0500 (Tue, 12 May 2009) | 6 lines Avoid initializing routines if the authentication fails. Fixes a crash (RR) issue. (closes issue #14508) Reported by: tiziano Patches: 20090221_2_wrongmailbox.diff.txt uploaded by tiziano (license 377) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193956 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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56275abb32 |
Update spiral support in trunk and 1.6.X to match what is in 1.4.
In 1.4, a SIP spiral is treated the same way as a call forward. This works much better than what is currently in trunk and 1.6.X. The code in trunk and 1.6.X did not create a new call to the recipient of the spiral, instead trying to continue the same call. In addition to just being plain wrong, this also had the side effect of only being able to spiral calls to other SIP channels. With this in place, as long as call forwards are honored, SIP spirals will work properly. This means that it will work for outbound calls made by the Queue, Dial, and Page applications. For originated calls and spool calls, however, the spiral will not work properly until a generic call forward mechanism is introduced into Asterisk. (relates to issue #13630) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193954 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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71cced338b |
Blocked revisions 193880 via svnmerge
........ r193880 | mmichelson | 2009-05-12 13:18:44 -0500 (Tue, 12 May 2009) | 12 lines Set the invitestate to INV_CANCELLED only if we are actually sending a SIP CANCEL. The problem was that the hangup code was setting the invitestate too early. The result of this was that we would always send a CANCEL request, even if it was not an appropriate time to do so (e.g. we have not yet received a provisional response for our INVITE). Note that this same fix had been applied to trunk and the 1.6.X branches starting with revision 155467. This is why you will see this revision being blocked from those places. AST-216 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193886 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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39ed871d95 |
Convert a THREADSTORAGE object into a simple malloc'd object (as suggested by Russell on -dev)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193870 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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1c988d8996 |
add 'const' qualifiers in various places where they should have been
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193832 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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59e8829e36 |
Found and fixed a memory leak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193757 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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76d7462a1f |
Recorded merge of revisions 193755 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193755 | tilghman | 2009-05-11 17:48:20 -0500 (Mon, 11 May 2009) | 18 lines Move 300 bytes around on the stack, to make more room for an extension buffer. This allows more concurrent extensions to be copied for a single voicemail, without creating a possibility of upsetting existing users, where a dialplan could run out of stack space where it had run fine before. Alternatively, we could have allocated off the heap, but that is a larger change and would have increased the chance for instability introduced by this change. This is really solved starting in 1.6.0.11, as the use of an ast_str buffer allows an unlimited number of extensions (up to available memory). We additionally create a new warning message when the buffer length is exceeded, permitting administrators to see an issue after the fact, whereas previously the list was silently truncated. (closes issue #14739) Reported by: p_lindheimer Patches: 20090417__bug14739.diff.txt uploaded by tilghman (license 14) Tested by: p_lindheimer ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193756 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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174697b7d1 |
Fix some timer state corruption.
In res_timer_timerfd, handle the case that set_rate gets called while a timer is still in continuous mode. In this case, we want to remember the configured rate, but not actually set it until continuous mode has been disabled. Thanks to dvossel for finding and helping to debug the problem. (closes issue #15080) Reported by: dvossel Tested by: dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193718 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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52541f5aeb |
Don't nullify an ast_str pointer.
(closes issue #15061) Reported by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193678 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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3b01ff719c |
Merged revisions 193613 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193613 | rmudgett | 2009-05-11 14:09:00 -0500 (Mon, 11 May 2009) | 12 lines Sent wrong message to clear a call we started if the other end has not responed yet. In the state MISDN_CALLING (i.e. SETUP was sent but no answer has arrived yet), it is not allowed to clear the call with RELEASE_COMPLETE. It must be cleared with DISCONNECT. A RELEASE_COMPLETE is only allowed as an answer to a SETUP. (See Q.931 ch. 5.3.2, 5.3.2.a, 5.3.2.b) Patches: chan-misdn-ccstate7.patch uploaded by customer. JIRA ABE-1862 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193614 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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9408242796 |
Recorded merge of revisions 193544 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193544 | lmadsen | 2009-05-11 13:35:17 -0400 (Mon, 11 May 2009) | 7 lines Document CHANNEL(transfercapability) in CLI documentation. (issue #15073) Reported by: pkempgen Patches: 20090511__issue15073.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193545 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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d0eab1d9a6 |
Fix a bug where receiving a control frame of subclass -1 would cause certain channels to get hung up.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193502 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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9c16774cc2 |
Minor documentation update for ast_event_queue().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193461 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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7e350686d6 |
Declare private data as static.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193459 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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fbad7a508d |
TCP not matching valid peer.
find_peer() does not find a valid peer when using pvt->recv as the sockaddr_in argument. Because of the way TCP works, the port number in pvt->recv is not what we're looking for at all. There is currently only one place that find_peer searches for a peer using the sockaddr_in argument. If the peer is not found after using pvt->recv (works for UDP since the port number will be correct), a temp sockaddr_in struct is made using the Contact header in the sip_request. This has the correct port number in it. Review: http://reviewboard.digium.com/r/236/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193387 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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7a2a6a073f |
Reset the members' call counts when resetting queue statistics.
This helps to prevent odd scenarios where a queue will claim to have taken 0 calls, but the members appear to have taken a non-zero amount. (closes issue #15068) Reported by: sum Patches: patchreset.patch uploaded by sum (license 766) Tested by: sum git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193349 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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0595e95a71 |
Fix the spelling of UNAVAILABLE in func_devstate CLI completion.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193274 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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86d63dc261 |
Merged revisions 193262 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193262 | dvossel | 2009-05-08 09:51:09 -0500 (Fri, 08 May 2009) | 9 lines "misdn show config" segfaults asterisk, if no MSN lists (closes issue #14976) Reported by: alecdavis Patches: misdn_config.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis, FabienToune ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193263 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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7893ab8fe7 |
Merged revisions 193193 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193193 | kpfleming | 2009-05-08 09:03:28 -0500 (Fri, 08 May 2009) | 7 lines Make absolute paths for logger channels work properly (Note: This is not a new feature, it was previously undocumented and broken.) The Asterisk logger has a feature to support absolute pathnames for logger channels, but the code implementing the feature was broken. This has been fixed, and the absolute path feature is now documented in the sample logger.conf. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193194 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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9cd0a94aeb |
Merged revisions 193119 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193119 | tilghman | 2009-05-07 18:41:11 -0500 (Thu, 07 May 2009) | 19 lines Fix Background within a Macro for FreePBX. If the single digit DTMF is an extension in the specified context, then go there and signal no DTMF. Otherwise, we should exit with that DTMF. If we're in Macro, we'll exit and seek that DTMF as the beginning of an extension in the Macro's calling context. If we're not in Macro, then we'll simply seek that extension in the calling context. Previously, someone complained about the behavior as it related to the interior of a Gosub routine, and the fix (#14011) inadvertently broke FreePBX (#14940). This change should fix both of these situations, but with the possible incompatibility that if a single digit extension does not exist (but a longer extension COULD have matched), it would have previously gone immediately to the "i" extension, but will now need to wait for a timeout. (closes issue #14940) Reported by: p_lindheimer Patches: 20090420__bug14940.diff.txt uploaded by tilghman (license 14) Tested by: p_lindheimer ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193120 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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90f76fcfba |
Merged revisions 193050 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193050 | rmudgett | 2009-05-07 17:17:06 -0500 (Thu, 07 May 2009) | 5 lines Give a more helpful message when an incoming call's dialed extension does not match. Added the dialed extension and context to the chan_misdn messages warning that the dialed number cannot be matched in the dialplan. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193077 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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e346cbb9bc |
Second result should not contain data from the first result.
(closes issue #15039) Reported by: jims Patches: 20090506__issue15039.diff.txt uploaded by tilghman (license 14) Tested by: jims git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193006 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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e4149506a1 |
Send DTMF frame before playing back audio.
(closes issue #14858) Reported by: barryf Patches: 20090507__bug14858.diff.txt uploaded by tilghman (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192938 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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01e5a86e1a |
Merged revisions 192932 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192932 | tilghman | 2009-05-07 11:29:08 -0500 (Thu, 07 May 2009) | 10 lines Eliminate repetition of fullcontact during reconstruction. If the fullcontact field appears in both the sippeers and the sipregs table, then during reconstruction of the field, it will otherwise be doubled. (closes issue #14754) Reported by: Alexei Gradinari Patches: 20090506__bug14754.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192933 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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1c05448f5a |
Merged revisions 192858 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192858 | jpeeler | 2009-05-06 17:15:19 -0500 (Wed, 06 May 2009) | 10 lines Make ParkedCall application stop execution of the dialplan after hang up Just changed park_exec to always return non-zero. I really wasn't entirely sure at first if this was a bug. Decided it was since it would be surprising when not using ParkedCall in the dialplan to hang up and have dialplan execution continue. (closes issue #14555) Reported by: francesco_r ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192861 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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658f81cb57 |
If no extension was found in the pattern tree, don't crash.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192853 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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9936f0ca14 |
Fix a bug where a timer would be created but not acknowledged.
This scenario crept up if chan_iax2 was loaded with no configuration file present. It would create a timer and tell it to go at an interval but the thread that normally acknowledges it would not be created because no configuration file was present. The timer will now be closed if no configuration file is present. (closes issue #15014) Reported by: madkins git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192808 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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829807cd2c |
Add numbers in Urdu, the national language of Pakistan
(closes issue #15034) Reported by: nasirq Patches: ast_say_number_full_ur-patch.c uploaded by nasirq (license 772) urdu.ods uploaded by nasirq (license 772) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192772 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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4d840c93b6 |
Make the code that prevents an infinite loop from happening into a case insensitive check.
(thanks eliel) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192736 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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38a5f51006 |
Fix an infinite loop with tab completion of CLI aliases that reference themselves.
(closes issue #15020) Reported by: junky git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192700 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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19916d118d |
Merged revisions 192633 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192633 | file | 2009-05-06 10:30:51 -0300 (Wed, 06 May 2009) | 7 lines Update some old logic to stop both begin and end DTMF frames from reaching the core if rfc2833 is not enabled. (closes issue #15036) Reported by: dimas Patches: v1-15036.patch uploaded by dimas (license 88) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192634 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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7019ff68db |
Fixed crashes from issue8824 review board channel locking changes.
The local struct ast_party_connected_line connected_caller variable was uninitialized when the copy function was called. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192590 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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549740d440 |
Merged revisions 192524 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192524 | seanbright | 2009-05-05 15:56:11 -0400 (Tue, 05 May 2009) | 11 lines Fix Javascript error when using astman.js in Internet Explorer. Internet Explorer (tested with 7.0) does not like trailing commas on constructs like object initializers, so get rid of them to avoid some errors. (closes issue #15026) Reported by: rajnishgiri Patches: bug15026.patch uploaded by seanbright (license 71) Tested by: seanbright ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192525 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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77e9d51c93 |
Merged revisions 192454 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192454 | file | 2009-05-05 15:22:27 -0300 (Tue, 05 May 2009) | 8 lines Fix an incorrect assumption that certain values on the channel will always exist when they may not. The CDR code involved with bridges wrongly assumed that the currently executing application and data values will always exist. It is possible for this to be false when call forwarding is involved. (closes issue #14984) Reported by: gincantalupo ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192462 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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1ed5422fa9 |
Merged revisions 192429 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192429 | file | 2009-05-05 14:43:30 -0300 (Tue, 05 May 2009) | 5 lines Fix a bug where the followme application would continue trying numbers after the caller hung up. (closes issue #13624) Reported by: sgenyuk ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192430 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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965b0f328e |
Revert CPC patch for now, until I decide whether or not it all should be merged into libss7/1.0 (It's still in the bug13495 branch and in libss7/trunk)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192427 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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2e7c1e3613 |
Fix a bug with setting t38pt_udptl at the user or peer level.
If an incoming call authenticated as a user or peer and t38pt_udptl was not set to yes in general then no UDPTL session would be present and any T38 related things would fail. This commit changes it so that if after authenticating T38 is enabled but no UDPTL session is present one will be created. (issue AST-215) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192387 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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2746f589b7 |
Add a more efficient way of allocating structures that use stringfields
This commit adds an API call that can be used to allocate a structure along with this stringfield storage in a single allocation. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192362 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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1f49e675bb |
Correct some flaws in the memory accounting code for stringfields and ao2 objects
Under some conditions, the memory allocation for stringfields and ao2 objects would not have supplied valid file/function names for MALLOC_DEBUG tracking, so this commit corrects that. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192357 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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ec5116f80c |
Properly account for memory allocated for channels and datastores
As in previous commits, when channels are allocated (with ast_channel_alloc) or datastores are allocated (with ast_datastore_alloc) properly account for the memory being owned by the caller, instead of the allocator function itself. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192318 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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d8182202ef |
Ensure that string pools allocated to hold stringfields are properly accounted in MALLOC_DEBUG mode
This commit modifies the stringfield pool allocator to remember the 'owner' of the stringfield manager the pool is being allocated for, and ensures that pools allocated in the future when fields are populated are owned by that file/function. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192279 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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0d44a84870 |
Merged revisions 192213 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192213 | dvossel | 2009-05-04 17:37:31 -0500 (Mon, 04 May 2009) | 11 lines global mohinterpret setting is ignored mohinterpret and mohsuggest global variables were not copied over during build_users and build_peers. (closes issue #14728) Reported by: dimas Patches: v1-14728.patch uploaded by dimas (license 88) Tested by: dimas, dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192214 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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e0aba74fa9 |
Restore 'asyncagi break' command to 1.6.1 and higher.
(closes issue #14985) Reported by: nikkk Patches: 20090428__bug14985.diff.txt uploaded by tilghman (license 14) 20090429__bug14985__1.6.1.diff.txt uploaded by tilghman (license 14) Tested by: nikkk git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192171 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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f69d6d46f1 |
Pass libraries in LIBS, not LDFLAGS.
(closes issue #14671) Reported by: Chainsaw Patches: asterisk-1.6.0.6-toolcheck-libs-not-ldflags.patch uploaded by Chainsaw (license 723) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192132 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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17dad4e5b0 |
Commit documentation changes related to issue #14801.
(issue #14801) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192096 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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1726475a54 |
Ensure that astobj2 memory allocations are properly accounted for when MALLOC_DEBUG is used
This commit ensures that all astobj2 allocated objects are properly accounted for in MALLOC_DEBUG mode by passing down the file/function/line information from the module/function that actually called the astobj2 allocation function. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192059 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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150edbb67d |
Do not re-define _POSIX_C_SOURCE if it was already defined.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192032 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |