mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
1967 Commits (dcefa607e53ecd23f201bf55bbb8f2c90a0510c8)
| Author | SHA1 | Message | Date |
|---|---|---|---|
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|
6d177693e0 |
Merged revisions 209098 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r209098 | dbrooks | 2009-07-27 11:33:50 -0500 (Mon, 27 Jul 2009) | 6 lines Fixing typos. Replaces "recieved" with "received" and "initilize" with "initialize" (closes issue #15571) Reported by: alecdavis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@209221 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
881cdb00e5 |
Merged revisions 209132 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r209132 | mmichelson | 2009-07-27 12:50:04 -0500 (Mon, 27 Jul 2009) | 24 lines Merged revisions 209131 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r209131 | mmichelson | 2009-07-27 12:44:06 -0500 (Mon, 27 Jul 2009) | 18 lines Allow for UDPTL to use only even-numbered ports if desired. There are some VoIP providers out there that will not accept SDP offers with odd numbered UDPTL ports. While it is my personal opinion that these VoIP providers are misinterpreting RFC 2327, it really is not a big deal to play along with their silly little games. Of course, since restricting UDPTL ports to only even numbers reduces the range of available ports by half, so the option to use only even port numbers is off by default. A user can enable the behavior by setting use_even_ports=yes in udptl.conf. (closes issue #15182) Reported by: CGMChris Patches: 15182.patch uploaded by mmichelson (license 60) Tested by: CGMChris ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@209133 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
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ab117ba86d |
Merged revisions 208924 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r208924 | jpeeler | 2009-07-26 20:20:37 -0500 (Sun, 26 Jul 2009) | 9 lines Merged revisions 208923 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208923 | jpeeler | 2009-07-26 20:18:31 -0500 (Sun, 26 Jul 2009) | 2 lines Fix logic errors from 208746 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208925 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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bf0a2c9fa5 |
Merged revisions 208749 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r208749 | jpeeler | 2009-07-25 01:23:18 -0500 (Sat, 25 Jul 2009) | 13 lines Merged revisions 208746 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208746 | jpeeler | 2009-07-25 01:19:50 -0500 (Sat, 25 Jul 2009) | 7 lines Fix compiling under dev-mode with gcc 4.4.0. Mostly trivial changes, but I did not know of any other way to fix the "dereferencing type-punned pointer will break strict-aliasing rules" error without creating a tmp variable in chan_skinny. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208752 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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f43a65fd21 |
Merged revisions 208548 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r208548 | kpfleming | 2009-07-24 10:02:53 -0500 (Fri, 24 Jul 2009) | 8 lines Resolve a T.38 negotiation issue left over from the udptl-updates merge. The udptl-updates branch that was merged yesterday failed to properly send back T.38 SDP responses with the correct error correction mode, if the incoming SDP from the other end caused us to change error correction modes. This patch corrects that situation. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208549 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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791d4f0478 |
Merged revisions 208464 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r208464 | kpfleming | 2009-07-23 16:57:24 -0500 (Thu, 23 Jul 2009) | 46 lines Rework of T.38 negotiation and UDPTL API to address interoperability problems Over the past couple of months, a number of issues with Asterisk negotiating (and successfully completing) T.38 sessions with various endpoints have been found. This patch attempts to address many of them, primarily focused around ensuring that the endpoints' MaxDatagram size is honored, and in addition by ensuring that T.38 session parameter negotiation is performed correctly according to the ITU T.38 Recommendation. The major changes here are: 1) T.38 applications in Asterisk (app_fax) only generate/receive IFP packets, they do not ever work with UDPTL packets. As a result of this, they cannot be allowed to generate packets that would overflow the other endpoints' MaxDatagram size after the UDPTL stack adds any error correction information. With this patch, the application is told the maximum *IFP* size it can generate, based on a calculation using the far end MaxDatagram size and the active error correction mode on the T.38 session. The same is true for sending *our* MaxDatagram size to the remote endpoint; it is computed from the value that the application says it can accept (for a single IFP packet) combined with the active error correction mode. 2) All treatment of T.38 session parameters as 'capabilities' in chan_sip has been removed; these parameters are not at all like audio/video stream capabilities. There are strict rules to follow for computing an answer to a T.38 offer, and chan_sip now follows those rules, using the desired parameters from the application (or channel) that wants to accept the T.38 negotiation. 3) chan_sip now stores and forwards ast_control_t38_parameters structures for tracking 'our' and 'their' T.38 session parameters; this greatly simplifies negotiation, especially for pass-through calls. 4) Since T.38 negotiation without specifying parameters or receiving the final negotiated parameters is not very worthwhile, the AST_CONTROL_T38 control frame has been removed. A note has been added to UPGRADE.txt about this removal, since any out-of-tree applications that use it will no longer function properly until they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review: https://reviewboard.asterisk.org/r/310/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@208468 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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51101e961b |
Merged revisions 207723 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r207723 | mmichelson | 2009-07-21 09:29:40 -0500 (Tue, 21 Jul 2009) | 11 lines Merged revisions 207714 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207714 | mmichelson | 2009-07-21 09:26:00 -0500 (Tue, 21 Jul 2009) | 5 lines Document default timeout for AMI originations. AST-224 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@207725 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 207680 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r207680 | kpfleming | 2009-07-21 08:28:04 -0500 (Tue, 21 Jul 2009) | 18 lines Merged revisions 207647 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207647 | kpfleming | 2009-07-21 08:04:44 -0500 (Tue, 21 Jul 2009) | 12 lines Ensure that user-provided CFLAGS and LDFLAGS are honored. This commit changes the build system so that user-provided flags (in ASTCFLAGS and ASTLDFLAGS) are supplied to the compiler/linker *after* all flags provided by the build system itself, so that the user can effectively override the build system's flags if desired. In addition, ASTCFLAGS and ASTLDFLAGS can now be provided *either* in the environment before running 'make', or as variable assignments on the 'make' command line. As a result, the use of COPTS and LDOPTS is no longer necessary, so they are no longer documented, but are still supported so as not to break existing build systems that supply them when building Asterisk. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@207683 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 207361 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r207361 | russell | 2009-07-20 11:36:15 -0500 (Mon, 20 Jul 2009) | 16 lines Merged revisions 207360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207360 | russell | 2009-07-20 11:26:24 -0500 (Mon, 20 Jul 2009) | 9 lines Only do the chan->fdno check in ast_read() in a developer build. I changed this check to only happen in a dev-mode build. I also added a comment explaining what is going on. I also made it so that detection of this situation does not affect ast_read() operation. (closes issue #14723) Reported by: seadweller ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@207362 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 145293,158010 from
https://origsvn.digium.com/svn/asterisk/branches/1.4 to make merging easier. These changes are already on trunk. ................ r145293 | rmudgett | 2008-09-30 18:55:24 -0500 (Tue, 30 Sep 2008) | 54 lines channels/chan_misdn.c channels/misdn/isdn_lib.c * Miscellaneous other fixes from trunk to make merging easier later. ........ r145200 | rmudgett | 2008-09-30 16:00:54 -0500 (Tue, 30 Sep 2008) | 7 lines * Miscellaneous formatting changes to make v1.4 and trunk more merge compatible in the mISDN area. channels/chan_misdn.c * Eliminated redundant code in cb_events() EVENT_SETUP ........ r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008) | 9 lines improved helptext of misdn_set_opt. ........ r142181 | rmudgett | 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) | 1 line Cleaned up comment ........ r138738 | rmudgett | 2008-08-18 16:07:28 -0500 (Mon, 18 Aug 2008) | 30 lines channels/chan_misdn.c * Made bearer2str() use allowed_bearers_array[] * Made use the causes.h defines instead of hardcoded numbers. * Made use Asterisk presentation indicator values if either of the mISDN presentation or screen options are negative. * Updated the misdn_set_opt application option descriptions. * Renamed the awkward Caller ID presentation misdn_set_opt application option value not_screened to restricted. Deprecated the not_screened option value. channels/misdn/isdn_lib.c * Made use the causes.h defines instead of hardcoded numbers. * Fixed some spelling errors and typos. * Added all defined facility code strings to fac2str(). channels/misdn/isdn_lib.h * Added doxygen comments to struct misdn_bchannel. channels/misdn/isdn_lib_intern.h * Added doxygen comments to struct misdn_stack. channels/misdn_config.c configs/misdn.conf.sample * Updated the mISDN presentation and screen parameter descriptions. doc/misdn.txt (doc/tex/misdn.tex) * Updated the misdn_set_opt application option descriptions. * Fixed some spelling errors and typos. ................ r158010 | rmudgett | 2008-11-19 19:46:09 -0600 (Wed, 19 Nov 2008) | 9 lines Merged revision 157977 from https://origsvn.digium.com/svn/asterisk/team/group/issue8824 ........ Fixes JIRA ABE-1726 The dial extension could be empty if you are using MISDN_KEYPAD to control ISDN provider features. ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@207286 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 206868 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r206868 | dvossel | 2009-07-16 16:25:22 -0500 (Thu, 16 Jul 2009) | 14 lines Merged revisions 206867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206867 | dvossel | 2009-07-16 16:24:16 -0500 (Thu, 16 Jul 2009) | 8 lines avoid segfault caused by user error If the CALLERPRES() dialplan function is set to nothing, a segfault occurs. This is user error to begin with, but I'd rather see a cli warning message than have Asterisk crash on me. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@206871 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 205939 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r205939 | kpfleming | 2009-07-10 13:44:09 -0500 (Fri, 10 Jul 2009) | 1 line Update comments about the level of T.38 support in Asterisk. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205940 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 205479 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r205479 | dvossel | 2009-07-08 18:19:09 -0500 (Wed, 08 Jul 2009) | 16 lines Merged revisions 205471 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205471 | dvossel | 2009-07-08 18:15:54 -0500 (Wed, 08 Jul 2009) | 10 lines Fixes 8khz assumptions Many calculations assume 8khz is the codec rate. This is not always the case. This patch only addresses chan_iax.c and res_rtp_asterisk.c, but I am sure there are other areas that make this assumption as well. Review: https://reviewboard.asterisk.org/r/306/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205597 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 205532 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r205532 | mvanbaak | 2009-07-09 10:31:24 +0200 (Thu, 09 Jul 2009) | 5 lines pthread_self returns a pthread_t which is not an unsigned int on all pthread implementations. Casting it to an unsigned int fixes compiler warnings. Tested on OpenBSD and Linux both 32 and 64 bit ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205533 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 205412 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r205412 | dvossel | 2009-07-08 17:15:06 -0500 (Wed, 08 Jul 2009) | 12 lines Merged revisions 205409 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205409 | dvossel | 2009-07-08 16:35:12 -0500 (Wed, 08 Jul 2009) | 6 lines moving ast_devstate_to_extenstate to pbx.c from devicestate.c ast_devstate_to_extenstate belongs in pbx.c. This change fixes a compile time error with chan_vpb as well. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205415 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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oops, fixing build
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Merged revisions 205196 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r205196 | tilghman | 2009-07-08 11:27:50 -0500 (Wed, 08 Jul 2009) | 9 lines Merged revisions 205188 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205188 | tilghman | 2009-07-08 11:26:15 -0500 (Wed, 08 Jul 2009) | 2 lines Add redirection warnings for the invalid language codes previously removed. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205200 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 205151 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r205151 | russell | 2009-07-08 10:56:28 -0500 (Wed, 08 Jul 2009) | 2 lines Use tabs instead of spaces for indentation. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205152 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 205120 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r205120 | russell | 2009-07-08 10:17:19 -0500 (Wed, 08 Jul 2009) | 16 lines Move OpenSSL initialization to a single place, make library usage thread-safe. While doing some reading about OpenSSL, I noticed a couple of things that needed to be improved with our usage of OpenSSL. 1) We had initialization of the library done in multiple modules. This has now been moved to a core function that gets executed during Asterisk startup. We already link OpenSSL into the core for TCP/TLS functionality, so this was the most logical place to do it. 2) OpenSSL is not thread-safe by default. However, making it thread safe is very easy. We just have to provide a couple of callbacks. One callback returns a thread ID. The other handles locking. For more information, start with the "Is OpenSSL thread-safe?" question on the FAQ page of openssl.org. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@205139 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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c3b8c5ead3 |
Restore Hungarian (mistakenly removed during merge)
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Merged revisions 204948 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r204948 | kpfleming | 2009-07-06 08:38:29 -0500 (Mon, 06 Jul 2009) | 7 lines Improve handling of AST_CONTROL_T38 and AST_CONTROL_T38_PARAMETERS for non-T.38-capable channels. This change allows applications that request T.38 negotiation on a channel that does not support it to get the proper indication that it is not supported, rather than thinking that negotiation was started when it was not. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@204949 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 204710 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r204710 | dvossel | 2009-07-02 11:03:44 -0500 (Thu, 02 Jul 2009) | 21 lines Merged revisions 204681 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204681 | dvossel | 2009-07-02 10:05:57 -0500 (Thu, 02 Jul 2009) | 14 lines Improved mapping of extension states from combined device states. This fixes a few issues with incorrect extension states and adds a cli command, core show device2extenstate, to display all possible state mappings. (closes issue #15413) Reported by: legart Patches: exten_helper.diff uploaded by dvossel (license 671) Tested by: dvossel, legart, amilcar Review: https://reviewboard.asterisk.org/r/301/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@204754 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 204563 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r204563 | tilghman | 2009-06-30 15:41:04 -0500 (Tue, 30 Jun 2009) | 13 lines Merged revisions 204556 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204556 | tilghman | 2009-06-30 15:23:51 -0500 (Tue, 30 Jun 2009) | 6 lines More incorrect language codes, plus ensuring that regionalizations use the specified language, and not English for grammar. (closes issue #15022) Reported by: greenfieldtech Patches: 20090519__issue15022.diff.txt uploaded by tilghman (license 14) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@204581 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 204475 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r204475 | qwell | 2009-06-30 13:48:35 -0500 (Tue, 30 Jun 2009) | 9 lines Merged revisions 204474 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204474 | qwell | 2009-06-30 13:47:06 -0500 (Tue, 30 Jun 2009) | 1 line Fix ast_say_counted_noun to correctly handle Polish. Fix a comment typo in passing. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@204476 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Recorded merge of revisions 204470 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r204470 | tilghman | 2009-06-30 13:36:24 -0500 (Tue, 30 Jun 2009) | 18 lines Recorded merge of revisions 204469 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204469 | tilghman | 2009-06-30 13:23:35 -0500 (Tue, 30 Jun 2009) | 11 lines "tw" is the language specification for Twi (from Ghana) not Taiwanese. (closes issue #15346) Reported by: volivier Patches: 20090617__issue15346__1.4.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__trunk.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__1.6.0.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__1.6.1.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__1.6.2.diff.txt uploaded by tilghman (license 14) Tested by: volivier ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@204471 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 203802 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r203802 | russell | 2009-06-26 16:21:48 -0500 (Fri, 26 Jun 2009) | 22 lines Merged revisions 203785 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203785 | russell | 2009-06-26 16:16:39 -0500 (Fri, 26 Jun 2009) | 15 lines Don't fast forward past the end of a message. This is nice change for users of the voicemail application. If someone gets a little carried away with fast forwarding through a message, they can easily get to the end and accidentally exit the voicemail application by hitting the fast forward key during the following prompt. This adds some safety by not allowing a fast forward past the end of a message. (closes issue #14554) Reported by: lacoursj Patches: 21761.patch uploaded by lacoursj (license 707) Tested by: lacoursj ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@203818 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 203699 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r203699 | file | 2009-06-26 16:27:24 -0300 (Fri, 26 Jun 2009) | 2 lines Improve T.38 negotiation by exchanging session parameters between application and channel. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@203701 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 203444 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r203444 | dvossel | 2009-06-25 16:45:32 -0500 (Thu, 25 Jun 2009) | 4 lines fixes a few redundant conditions (issue #15269) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@203447 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 203381 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r203381 | twilson | 2009-06-25 16:15:11 -0500 (Thu, 25 Jun 2009) | 11 lines Merged revisions 203380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203380 | twilson | 2009-06-25 16:13:10 -0500 (Thu, 25 Jun 2009) | 4 lines I didn't see that Mark already fixed the underlying issue! Yay for removing useless code. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@203387 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 203376 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r203376 | russell | 2009-06-25 16:04:55 -0500 (Thu, 25 Jun 2009) | 16 lines Merged revisions 203375 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203375 | russell | 2009-06-25 16:02:18 -0500 (Thu, 25 Jun 2009) | 9 lines Fix a case where CDR answer time could be before the start time involving parking. (closes issue #13794) Reported by: davidw Patches: 13794.patch uploaded by murf (license 17) 13794.patch.160 uploaded by murf (license 17) Tested by: murf, dbrooks ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@203377 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 203231 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r203231 | mmichelson | 2009-06-25 13:53:12 -0500 (Thu, 25 Jun 2009) | 8 lines Blocked revisions 203230 via svnmerge ........ r203230 | mmichelson | 2009-06-25 13:52:22 -0500 (Thu, 25 Jun 2009) | 3 lines Prevent false positives when freeing a NULL pointer with MALLOC_DEBUG enabled. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@203232 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 202753 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r202753 | rbrindley | 2009-06-23 16:25:17 -0500 (Tue, 23 Jun 2009) | 9 lines If we delete the info, lets also delete the lines (closes issue #14509) Reported by: timeshell Patches: 20090504__bug14509.diff.txt uploaded by tilghman (license 14) Tested by: awk, timeshell ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@202754 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 202497 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r202497 | russell | 2009-06-22 15:11:04 -0500 (Mon, 22 Jun 2009) | 11 lines Merged revisions 202496 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r202496 | russell | 2009-06-22 15:08:53 -0500 (Mon, 22 Jun 2009) | 4 lines Report CallerID change during a masquerade. Reported by: markster ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@202498 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 201829 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r201829 | tilghman | 2009-06-18 19:43:41 -0500 (Thu, 18 Jun 2009) | 13 lines Merged revisions 201828 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201828 | tilghman | 2009-06-18 19:40:41 -0500 (Thu, 18 Jun 2009) | 6 lines If the "h" extension fails, give it another chance in main/pbx.c. If the "h" extension fails, give it another chance in main/pbx.c, when it returns from the bridge code. Fixes an issue where the "h" extension may occasionally not fire, when a Dial is executed from a Macro. Debugged in #asterisk with user tompaw. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@201830 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 201678 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r201678 | dvossel | 2009-06-18 11:37:42 -0500 (Thu, 18 Jun 2009) | 11 lines fixes some memory leaks and redundant conditions (closes issue #15269) Reported by: contactmayankjain Patches: patch.txt uploaded by contactmayankjain (license 740) memory_leak_stuff.trunk.diff uploaded by dvossel (license 671) Tested by: contactmayankjain, dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@201682 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 201458 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r201458 | mmichelson | 2009-06-17 15:04:12 -0500 (Wed, 17 Jun 2009) | 15 lines Merged revisions 201450 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201450 | mmichelson | 2009-06-17 14:59:31 -0500 (Wed, 17 Jun 2009) | 9 lines Change the datastore traversal in ast_do_masquerade to use a safe list traversal. It is possible for datastore fixup functions to remove the datastore from the list and free it. In particular, the queue_transfer_fixup in app_queue does this. While I don't yet know of this causing any crashes, it certainly could. Found while discussing a separate issue with Brian Degenhardt. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@201459 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 201056,201090 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r201056 | kpfleming | 2009-06-16 13:54:30 -0500 (Tue, 16 Jun 2009) | 18 lines Merged revisions 200991 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun 2009) | 11 lines Improve support for media paths that can generate multiple frames at once. There are various media paths in Asterisk (codec translators and UDPTL, primarily) that can generate more than one frame to be generated when the application calling them expects only a single frame. This patch addresses a number of those cases, at least the primary ones to solve the known problems. In addition it removes the broken TRACE_FRAMES support, fixes a number of bugs in various frame-related API functions, and cleans up various code paths affected by these changes. https://reviewboard.asterisk.org/r/175/ ........ ................ r201090 | kpfleming | 2009-06-16 14:27:12 -0500 (Tue, 16 Jun 2009) | 5 lines Another minor fix to compiler attribute checking. Defaulting to 'static' for the function scope was bad... so remove it. ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@201093 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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660bceff3c |
Merged revisions 200361 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r200361 | mmichelson | 2009-06-12 14:07:51 -0500 (Fri, 12 Jun 2009) | 16 lines Merged revisions 200360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200360 | mmichelson | 2009-06-12 14:06:41 -0500 (Fri, 12 Jun 2009) | 10 lines Suppress a warning message and give a better return code when generating inband ringing after a call is answered. (closes issue #15158) Reported by: madkins Patches: 15158.patch uploaded by mmichelson (license 60) Tested by: madkins ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@200362 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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3516eaa7e0 |
Fixes the argument order in definition of new_find_extension().
In the definition of new_find_extension(), the arguments 'callerid' and 'label' were swapped. The prototype declaration and all calls to the function are ordered 'callerid' then 'label', but the function itself was ordered 'label' then 'callerid'. (closes issue #15303) Reported by: JimDickenson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@199994 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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8e3df8bd1f |
Merged revisions 199298 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r199298 | dvossel | 2009-06-05 16:21:22 -0500 (Fri, 05 Jun 2009) | 21 lines Merged revisions 199297 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199297 | dvossel | 2009-06-05 16:19:56 -0500 (Fri, 05 Jun 2009) | 14 lines Fixes issue with hints giving unexpected results. Hints with two or more devices that include ONHOLD gave unexpected results. (closes issue #15057) Reported by: p_lindheimer Patches: onhold_trunk.diff uploaded by dvossel (license 671) pbx.c.1.4.patch uploaded by p (license 558) devicestate.c.trunk.patch uploaded by p (license 671) Tested by: p_lindheimer, dvossel Review: https://reviewboard.asterisk.org/r/254/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@199301 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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8340a34a4a |
Merged revisions 199051 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r199051 | seanbright | 2009-06-04 10:31:24 -0400 (Thu, 04 Jun 2009) | 47 lines Merged revisions 199022 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun 2009) | 40 lines Safely handle AMI connections/reload requests that occur during startup. During asterisk startup, a lock on the list of modules is obtained by the primary thread while each module is initialized. Issue 13778 pointed out a problem with this approach, however. Because the AMI is loaded before other modules, it is possible for a module reload to be issued by a connected client (via Action: Command), causing a deadlock. The resolution for 13778 was to move initialization of the manager to happen after the other modules had already been lodaded. While this fixed this particular issue, it caused a problem for users (like FreePBX) who call AMI scripts via an #exec in a configuration file (See issue 15189). The solution I have come up with is to defer any reload requests that come in until after the server is fully booted. When a call comes in to ast_module_reload (from wherever) before we are fully booted, the request is added to a queue of pending requests. Once we are done booting up, we then execute these deferred requests in turn. Note that I have tried to make this a bit more intelligent in that it will not queue up more than 1 request for the same module to be reloaded, and if a general reload request comes in ('module reload') the queue is flushed and we only issue a single deferred reload for the entire system. As for how this will impact existing installations - Before 13778, a reload issued before module initialization was completed would result in a deadlock. After 13778, you simply couldn't connect to the manager during startup (which causes problems with #exec-that-calls-AMI configuration files). I believe this is a good general purpose solution that won't negatively impact existing installations. (closes issue #15189) (closes issue #13778) Reported by: p_lindheimer Patches: 06032009_15189_deferred_reloads.diff uploaded by seanbright (license 71) Tested by: p_lindheimer, seanbright Review: https://reviewboard.asterisk.org/r/272/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@199052 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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9c6652d306 |
Merged revisions 198856 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r198856 | dvossel | 2009-06-02 16:17:49 -0500 (Tue, 02 Jun 2009) | 10 lines Generic call forward api, ast_call_forward() The function ast_call_forward() forwards a call to an extension specified in an ast_channel's call_forward string. After an ast_channel is called, if the channel's call_forward string is set this function can be used to forward the call to a new channel and terminate the original one. I have included this api call in both channel.c's ast_request_and_dial() and feature.c's feature_request_and_dial(). App_dial and app_queue already contain call forward logic specific for their application and options. (closes issue #13630) Reported by: festr Review: https://reviewboard.asterisk.org/r/271/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@198889 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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95fac13256 |
Merged revisions 198072 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r198072 | mnicholson | 2009-05-29 14:04:24 -0500 (Fri, 29 May 2009) | 21 lines Merged revisions 198068 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198068 | mnicholson | 2009-05-29 13:53:01 -0500 (Fri, 29 May 2009) | 15 lines Use AST_CDR_NOANSWER instead of AST_CDR_NULL as the default CDR disposition. This change also involves the addition of an AST_CDR_FLAG_ORIGINATED flag that is used on originated channels to distinguish: them from dialed channels. (closes issue #12946) Reported by: meral Patches: null-cdr2.diff uploaded by mnicholson (license 96) Tested by: mnicholson, dbrooks (closes issue #15122) Reported by: sum Tested by: sum ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@198073 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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a66b938920 |
Merged revisions 197606 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r197606 | mmichelson | 2009-05-28 10:32:19 -0500 (Thu, 28 May 2009) | 22 lines Recorded merge of revisions 197588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197588 | mmichelson | 2009-05-28 10:27:49 -0500 (Thu, 28 May 2009) | 16 lines Allow for media to arrive from an alternate source when responding to a reinvite with 491. When we receive a SIP reinvite, it is possible that we may not be able to process the reinvite immediately since we have also sent a reinvite out ourselves. The problem is that whoever sent us the reinvite may have also sent a reinvite out to another party, and that reinvite may have succeeded. As a result, even though we are not going to accept the reinvite we just received, it is important for us to not have problems if we suddenly start receiving RTP from a new source. The fix for this is to grab the media source information from the SDP of the reinvite that we receive. This information is passed to the RTP layer so that it will know about the alternate source for media. Review: https://reviewboard.asterisk.org/r/252 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@197615 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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c7731d3489 |
Merged revisions 197543 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r197543 | mmichelson | 2009-05-28 09:58:06 -0500 (Thu, 28 May 2009) | 27 lines Merged revisions 197537 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197537 | mmichelson | 2009-05-28 09:49:13 -0500 (Thu, 28 May 2009) | 21 lines Add flags to chanspy audiohook so that audio stays in sync. There are two flags being added to the chanspy audiohook here. One is the pre-existing AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set, we ensure that the read and write slinfactories on the audiohook do not skew beyond a certain tolerance. In addition, there is a new audiohook flag added here, AST_AUDIOHOOK_SMALL_QUEUE. With this flag set, we do not allow for a slinfactory to build up a substantial amount of audio before flushing it. For this particular issue, this means that the person spying on the call will hear the conversations in real time with very little delay in the audio. (closes issue #13745) Reported by: geoffs Patches: 13745.patch uploaded by mmichelson (license 60) Tested by: snblitz ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@197544 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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5d28a72130 |
Merged revisions 197538 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r197538 | file | 2009-05-28 11:51:43 -0300 (Thu, 28 May 2009) | 5 lines Fix a bug in stringfields where it did not actually free the pools of memory. (closes issue #15074) Reported by: pj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@197539 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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76b9a6b5d8 |
Fix broken attended transfers
The bridge was terminating immediately after the attended transfer was completed. The problem was because upon reentering ast_channel_bridge nexteventts was checked to see if it was set and if so could possibly return AST_BRIDGE_COMPLETE. (closes issue #15183) Reported by: andrebarbosa Tested by: andrebarbosa, tootai, loloski git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@197126 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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e5d7298863 |
Merged revisions 195882 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r195882 | mnicholson | 2009-05-21 10:33:55 -0500 (Thu, 21 May 2009) | 20 lines Merged revisions 195881 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195881 | mnicholson | 2009-05-21 10:25:50 -0500 (Thu, 21 May 2009) | 13 lines This commit prevents cdr records with AST_CDR_FLAG_ANSLOCKED and AST_CDR_FLAG_LOCKED from being updated in certain cases. This is accomplished by adding two functions to update the answer time and disposition of calls that checks for the proper lock flags. These functions are used in the ast_bridge_call() function so that ForkCDR(A) calls are respected. This patch also modifies the way ast_bridge_call() chooses the cdr record to base the bridged_cdr on. Previously the first unlocked cdr record would be chosen, now instead the first cdr record is chosen and forked cdr records are moved to the bridge_cdr. This allows the original cdr record and any forked cdr records to be properly updated with answer and end times. (closes issue #13797) Reported by: sh0t Tested by: sh0t (closes issue #14744) Reported by: deepesh ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@195894 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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4c8cf3398e |
Merged revisions 195698 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r195698 | file | 2009-05-20 14:33:02 -0300 (Wed, 20 May 2009) | 12 lines Merged revisions 195688 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195688 | file | 2009-05-20 14:30:25 -0300 (Wed, 20 May 2009) | 5 lines Fix some code that wrongly assumed a pointer would always be non-NULL when dealing with CDRs after a bridge. (closes issue #15079) Reported by: barryf ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@195702 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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91b11ad2a3 |
Merged revisions 195369 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r195369 | eliel | 2009-05-18 16:49:20 -0400 (Mon, 18 May 2009) | 8 lines Fix the CLI command 'manager show command' documentation and functionality. The CLI command 'manager show command' supports passing multiple action names in the same line, but it was not allowing that because of a incorrect check in the argumentes counter. Also the documentation was updated to show that this usage of the command is possible. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@195442 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |