mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
18186 Commits (a076c9af215d04ae476d32fe2f8c32d4e652ea95)
| Author | SHA1 | Message | Date |
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a076c9af21 |
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/1.6.2@205413 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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aae9495def |
Merged revisions 205410 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r205410 | dvossel | 2009-07-08 17:02:54 -0500 (Wed, 08 Jul 2009) | 3 lines missing comma in devstatestring array ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@205411 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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88aad0a21e |
Merged revisions 205350 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r205350 | mmichelson | 2009-07-08 14:26:55 -0500 (Wed, 08 Jul 2009) | 20 lines Merged revisions 205349 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205349 | mmichelson | 2009-07-08 14:26:13 -0500 (Wed, 08 Jul 2009) | 14 lines Prevent phantom calls to queue members. If a caller were to hang up while a periodic announcement or position were being said, the return value for those functions would incorrectly indicate that the caller was still in the queue. With these changes, the problem does not occur. (closes issue #14631) Reported by: latinsud Patches: queue_announce_ghost_call2.diff uploaded by latinsud (license 745) (with small modification from me) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@205353 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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f357240896 |
Merged revisions 205291 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r205291 | qwell | 2009-07-08 13:19:46 -0500 (Wed, 08 Jul 2009) | 9 lines Merged revisions 205288 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205288 | qwell | 2009-07-08 13:19:03 -0500 (Wed, 08 Jul 2009) | 1 line Update config.guess and config.sub from the savannah.gnu.org git repo. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@205302 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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aef389ab58 |
Merged revisions 205254 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r205254 | dbrooks | 2009-07-08 12:26:26 -0500 (Wed, 08 Jul 2009) | 8 lines Fixes Park() argument handling Park() was not respecting the arguments passed to it. Any extension/context/priority given to it was being ignored. This patch remedies this. (closes issue #15380) Reported by: DLNoah ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@205287 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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c1daabaf2a |
oops, fixing build
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@205223 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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49ac045c97 |
Merged revisions 205216 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r205216 | dvossel | 2009-07-08 11:54:24 -0500 (Wed, 08 Jul 2009) | 17 lines Merged revisions 205215 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205215 | dvossel | 2009-07-08 11:53:40 -0500 (Wed, 08 Jul 2009) | 10 lines ast_samp2tv needs floating point for 16khz audio In ast_samp2tv(), (1000000 / _rate) = 62.5 when _rate is 16000. The .5 is currently stripped off because we don't calculate using floating points. This causes madness with 16khz audio. (issue ABE-1899) Review: https://reviewboard.asterisk.org/r/305/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@205217 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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599e2290fe |
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/1.6.2@205207 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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2d4a63ae59 |
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/1.6.2@205154 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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e97c471b2c |
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/1.6.2@205148 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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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/1.6.2@204951 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 204919 via svnmerge
........ r204919 | seanbright | 2009-07-03 11:44:01 -0400 (Fri, 03 Jul 2009) | 5 lines Add a configure check for Reverse Charging Indication support in LibPRI. Also go back and wrap all of the places that use the specific reverse charge APIs with preprocessor conditionals. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@204920 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 204835 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r204835 | rmudgett | 2009-07-02 17:01:28 -0500 (Thu, 02 Jul 2009) | 17 lines Merged revisions 204834 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204834 | rmudgett | 2009-07-02 16:59:43 -0500 (Thu, 02 Jul 2009) | 10 lines Removed confusing warning message "Got Busy in Connected State" If an incoming mISDN call is answered with the Answer application and a subsequent Dial gets a busy endpoint then it is valid for that already connected channel to get the busy indication. Asterisk will play the busy tones until the dialplan plays something else or hangs up the call. (closes issue #11974) Reported by: fvdb ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@204838 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 204749 via svnmerge
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r204749 | seanbright | 2009-07-02 13:46:14 -0400 (Thu, 02 Jul 2009) | 21 lines
Support setting and receiving Reverse Charging Indication over ISDN PRI.
This is a continuation of revision 885 to LibPRI (Capture and expose the Reverse
Charging Indication IE on ISDN PRI) which added the ability to get/set Reverse
Charging Indication in LibPRI. This patch adds the ability to specify RCI on
the outbound leg of a PRI call from within Asterisk, by prefixing the dialed
number with a capital 'C' like:
...,Dial(DAHDI/g1/C4445556666)
And to read it off an inbound channel:
exten => s,1,Set(RCI=${CHANNEL(reversecharge)})
Thanks again to rmudgett for the thorough review.
(closes issue #13760)
Reported by: mrgabu
Review: https://reviewboard.asterisk.org/r/303/
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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/1.6.2@204711 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 204622 via svnmerge
........ r204622 | seanbright | 2009-07-01 12:06:18 -0400 (Wed, 01 Jul 2009) | 2 lines A bunch of CODING_GUIDELINES related fixes. Not even close to done. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@204623 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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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/1.6.2@204611 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Blocked revisions 204561 via svnmerge
........ r204561 | seanbright | 2009-06-30 16:39:39 -0400 (Tue, 30 Jun 2009) | 1 line Remove an unnecessary #ifdef ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@204562 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Blocked revisions 204532 via svnmerge
........ r204532 | mmichelson | 2009-06-30 14:59:20 -0500 (Tue, 30 Jun 2009) | 5 lines Move the masquerade in local_attended_transfer to a point where we hold the channel lock. Masquerading without the channel's lock held is a *horrible* idea. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@204533 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Blocked revisions 204530 via svnmerge
........ r204530 | mmichelson | 2009-06-30 14:55:59 -0500 (Tue, 30 Jun 2009) | 10 lines Remove some bogus deadlock avoidance code from local_attended_transfer. First of all, the code was unnecessary. The goal was to lock a channel which was already locked. Second, the assumption of the deadlock avoidance loop was that the sip_pvt was already locked and we were trying to get the channel lock. The problem is that the sip_pvt was unlocked a few lines above. Basically, I'm removing 5 lines of no-op. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@204531 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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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/1.6.2@204478 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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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/1.6.2@204473 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 204440 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r204440 | russell | 2009-06-30 12:22:16 -0500 (Tue, 30 Jun 2009) | 2 lines Rename res_config_sqlite.conf to res_config_sqlite.conf.sample (missing .sample). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@204442 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 204422-204423,204428 via svnmerge
........ r204422 | russell | 2009-06-30 12:15:09 -0500 (Tue, 30 Jun 2009) | 2 lines Rename res_mysql.conf to res_config_mysql.conf, make module support both ........ r204423 | russell | 2009-06-30 12:16:56 -0500 (Tue, 30 Jun 2009) | 2 lines Rename mobile.conf to chan_mobile.conf, make module support old name, too ........ r204428 | russell | 2009-06-30 12:18:18 -0500 (Tue, 30 Jun 2009) | 2 lines Rename ooh323.conf to chan_ooh323.conf, make module support both names ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@204430 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 204417-204420 via svnmerge
........ r204417 | russell | 2009-06-30 12:08:14 -0500 (Tue, 30 Jun 2009) | 2 lines Rename app_addon_sql_mysql to app_mysql ........ r204418 | russell | 2009-06-30 12:09:04 -0500 (Tue, 30 Jun 2009) | 2 lines Rename cdr_addon_mysql to cdr_mysql ........ r204419 | russell | 2009-06-30 12:10:45 -0500 (Tue, 30 Jun 2009) | 2 lines Rename mysql.conf to app_mysql.conf, make module support both names ........ r204420 | russell | 2009-06-30 12:11:31 -0500 (Tue, 30 Jun 2009) | 2 lines Make addons build last - this is for Qwell. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@204421 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Blocked revisions 204415 via svnmerge
........ r204415 | kpfleming | 2009-06-30 12:04:35 -0500 (Tue, 30 Jun 2009) | 8 lines Add-ons related build system improvements. Ensure that add-on modules can be embedded, fix up Makefile.moddir_rules to allow module directory Makefiles to more easily specify the modules to be built, and explicitly list the addons modules in its Makefile, since the module names don't follow any pattern. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@204416 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 204413 via svnmerge
........ r204413 | russell | 2009-06-30 11:40:38 -0500 (Tue, 30 Jun 2009) | 12 lines Move Asterisk-addons modules into the main Asterisk source tree. Someone asked yesterday, "is there a good reason why we can't just put these modules in Asterisk?". After a brief discussion, as long as the modules are clearly set aside in their own directory and not enabled by default, it is perfectly fine. For more information about why a module goes in addons, see README-addons.txt. chan_ooh323 does not currently compile as it is behind some trunk API updates. However, it will not build by default, so it should be okay for now. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@204414 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 204355 via svnmerge
........ r204355 | seanbright | 2009-06-29 19:50:46 -0400 (Mon, 29 Jun 2009) | 2 lines A few const changes in app_meetme.c that I noticed while browsing the source. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@204356 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 204301 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r204301 | mmichelson | 2009-06-29 17:50:35 -0500 (Mon, 29 Jun 2009) | 15 lines Merged revisions 204300 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204300 | mmichelson | 2009-06-29 17:45:34 -0500 (Mon, 29 Jun 2009) | 9 lines Add error message so that it is clear why a SIP peer was not processed when a DNS lookup fails on a host or outboundproxy. (closes issue #13432) Reported by: p_lindheimer Patches: outboundproxy.patch uploaded by p (license 558) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@204304 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Merged revisions 204247 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r204247 | mmichelson | 2009-06-29 16:48:54 -0500 (Mon, 29 Jun 2009) | 32 lines Merged revisions 204243,204246 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204243 | mmichelson | 2009-06-29 16:23:43 -0500 (Mon, 29 Jun 2009) | 22 lines Fix a problem where chan_sip would ignore "old" but valid responses. chan_sip has had a problem for quite a long time that would manifest when Asterisk would send multiple SIP responses on the same dialog before receiving a response. The problem occurred because chan_sip only kept track of the highest outgoing sequence number used on the dialog. If Asterisk sent two requests out, and a response arrived for the first request sent, then Asterisk would ignore the response. The result was that Asterisk would continue retransmitting the requests and ignoring the responses until the maximum number of retransmissions had been reached. The fix here is to rearrange the code a bit so that instead of simply comparing the sequence number of the response to our latest outgoing sequence number, we walk our list of outstanding packets and determine if there is a match. If there is, we continue. If not, then we ignore the response. In doing this, I found a few completely useless variables that I have now removed. (closes issue #11231) Reported by: flefoll Review: https://reviewboard.asterisk.org/r/298 ........ r204246 | mmichelson | 2009-06-29 16:37:05 -0500 (Mon, 29 Jun 2009) | 3 lines Fix build oops. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@204250 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 204217 via svnmerge
........ r204217 | seanbright | 2009-06-29 16:29:10 -0400 (Mon, 29 Jun 2009) | 1 line Reorganize this adaptive CEL config a bit. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@204222 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 204119,204143 via svnmerge
........ r204119 | seanbright | 2009-06-29 14:05:27 -0400 (Mon, 29 Jun 2009) | 1 line Add common headers to CEL related configs. ........ r204143 | seanbright | 2009-06-29 14:44:44 -0400 (Mon, 29 Jun 2009) | 7 lines Get app_rpt compiling again. I doubt seriously that it actually works. Also, the code in this module is horrendous and we should remove it from the tree. I'm not sure who is supposed to be maintaning this thing, but they clearly are not. I don't see the sense of leaving it in the main tree. If it lives *anywhere* it should be in addons. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@204145 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 204013 via svnmerge
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r204013 | mmichelson | 2009-06-29 10:04:39 -0500 (Mon, 29 Jun 2009) | 11 lines
Blocked revisions 204012 via svnmerge
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r204012 | mmichelson | 2009-06-29 10:04:17 -0500 (Mon, 29 Jun 2009) | 6 lines
Place unlock of mutex in an else block so that it does not get unlocked twice.
(closes issue #15400)
Reported by: aragon
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Blocked revisions 203985 via svnmerge
........ r203985 | seanbright | 2009-06-27 16:26:01 -0400 (Sat, 27 Jun 2009) | 1 line Another CHANGES spelling fix. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@203986 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 203962 via svnmerge
........ r203962 | russell | 2009-06-27 05:04:51 -0500 (Sat, 27 Jun 2009) | 8 lines Only update total silence counter after a counter reset. (closes issue #2264) Reported by: pfn Patches: silent-vm-1.6.2-fix2.txt uploaded by pfn (license 810) Tested by: pfn ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@203963 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Merged revisions 203960 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r203960 | russell | 2009-06-27 04:51:45 -0500 (Sat, 27 Jun 2009) | 2 lines Minor tweaks and spelling fixes for CHANGES and UPGRADE.txt. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@203961 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Merged revisions 203909 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r203909 | rmudgett | 2009-06-26 20:07:52 -0500 (Fri, 26 Jun 2009) | 23 lines Merged revisions 203908 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203908 | rmudgett | 2009-06-26 19:55:12 -0500 (Fri, 26 Jun 2009) | 16 lines The ISDN CPE side should not exclusively pick B channels normally. Before this patch, Asterisk unconditionally picked B channels exclusively on the CPE side and normally allowed alternative B channels on the network side. Now Asterisk does the opposite. Reasons for the CPE side to normally not pick B channels exclusively: * For CPE point-to-multipoint mode (i.e. phone side), the CPE side does not have enough information to exclusively pick B channels. (There may be other devices on the line.) * Q.931 gives preference to the network side picking B channels. * Some telcos require the CPE side to not pick B channels exclusively. (closes issue #14383) Reported by: mbrancaleoni ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@203941 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 203853 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r203853 | jpeeler | 2009-06-26 17:11:31 -0500 (Fri, 26 Jun 2009) | 12 lines Merged revisions 203848 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r203848 | jpeeler | 2009-06-26 17:09:19 -0500 (Fri, 26 Jun 2009) | 5 lines Make sure to recreate the dahdi pseudo channel after dahdi restart (closes issue #14477) Reported by: timking ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@203857 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Blocked revisions 203846 via svnmerge
........ r203846 | seanbright | 2009-06-26 18:08:05 -0400 (Fri, 26 Jun 2009) | 14 lines Add a new module, cdr_syslog, which allows writing CDRs to syslog. The original patch for this was written by Brett Bryant, and I split it out into it's own module. (closes issue #12876) Reported by: bbryant Patches: 06162008_cdr_custom_syslog.diff uploaded by bbryant (license 36) 05212009_cdr_syslog.patch uploaded by seanbright (license 71) Tested by: seanbright Review: https://reviewboard.asterisk.org/r/297/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@203854 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Blocked revisions 203842 via svnmerge
........ r203842 | russell | 2009-06-26 16:48:41 -0500 (Fri, 26 Jun 2009) | 7 lines Add 's' option to ChanSpy, which makes the app exit when no channels are left to spy on. (closes issue #14594) Reported by: JimDickenson Patches: chanspy.diff uploaded by JimDickenson (license 710) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@203843 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/1.6.2@203828 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Blocked revisions 203783 via svnmerge
........ r203783 | mmichelson | 2009-06-26 15:52:19 -0500 (Fri, 26 Jun 2009) | 8 lines Add timestamp to response to "Ping" manager action. (closes issue #14596) Reported by: JimDickenson Patches: pong2.diff uploaded by JimDickenson (license 710) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@203784 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Merged revisions 203779 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r203779 | russell | 2009-06-26 15:45:00 -0500 (Fri, 26 Jun 2009) | 5 lines Ensure the TCP read buffer is fully initialized before handling each packet. (closes issue #14452) Reported by: umberto71 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@203782 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 203735 via svnmerge
........ r203735 | file | 2009-06-26 17:19:49 -0300 (Fri, 26 Jun 2009) | 6 lines Fix the 'nat' option to actually do RFC3581 as expected and extend the configurable values for finer control. (closes issue #8855) Reported by: mikma Tested by: klaus3000, file ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@203741 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 203721 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r203721 | dbrooks | 2009-06-26 15:13:51 -0500 (Fri, 26 Jun 2009) | 16 lines Fixing voicemail's error in checking max silence vs min message length Max silence was represented in milliseconds, yet vmminsecs (minmessage) was represented as seconds. Also, the inequality was reversed. The warning, if triggered, was "Max silence should be less than minmessage or you may get empty messages", which should have been logged if max silence was greater than minmessage, but the check was for less than. Also, conforming if statement to coding guidelines. closes issue #15331) Reported by: markd Review: https://reviewboard.asterisk.org/r/293/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@203731 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 203702 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r203702 | russell | 2009-06-26 14:31:14 -0500 (Fri, 26 Jun 2009) | 5 lines Make invalid hints report Unavailable instead of Idle. (closes issue #14413) Reported by: pj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@203715 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 203710 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r203710 | dvossel | 2009-06-26 14:47:11 -0500 (Fri, 26 Jun 2009) | 7 lines moving debug message from level 0 to 1. (closes issue #15404) Reported by: leobrown Patches: iax_codec_debug.patch uploaded by leobrown (license 541) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@203712 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 203672 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r203672 | jpeeler | 2009-06-26 14:03:25 -0500 (Fri, 26 Jun 2009) | 16 lines Check if polarityonanswerdelay has elapsed before setting a channel as answered after a polarity reversal. Previously on a polarity switch event chan_dahdi would set the channel immediately as answered. This would cause problems if a polarity reversal occurred when the line was picked up as the dial would not have yet occurred. Now if the polarity reversal occurs before delay has elapsed after coming off hook or an answer, it is ignored. Also, some refactoring was done in _handle_event. (closes issue #13917) Reported by: alecdavis Patches: chan_dahdi.bug13917.feb09.diff2.txt uploaded by alecdavis (license 585) Tested by: alecdavis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@203709 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/1.6.2@203705 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Blocked revisions 203638,203640 via svnmerge
........ r203638 | russell | 2009-06-26 10:28:53 -0500 (Fri, 26 Jun 2009) | 14 lines Merge the new Channel Event Logging (CEL) subsystem. CEL is the new system for logging channel events. This was inspired after facing many problems trying to represent what is possible to happen to a call in Asterisk using CDR records. For more information on CEL, see the built in HTML or PDF documentation generated from the files in doc/tex/. Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard work developing this code. Also, thanks to Matt Nicholson (mnicholson) and Sean Bright (seanbright) for their assistance in the final push to get this code ready for Asterisk trunk. Review: https://reviewboard.asterisk.org/r/239/ ........ r203640 | russell | 2009-06-26 10:42:26 -0500 (Fri, 26 Jun 2009) | 2 lines Note a new API call, and one that changed in doxygen. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@203641 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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