mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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23544 Commits (657aa491f00b7a3693ab8ddfb4623f516fda33b0)
Author | SHA1 | Message | Date |
---|---|---|---|
|
0d3dfad94f |
Merged revisions 379021,379023 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk ................ r379021 | dlee | 2013-01-14 09:29:22 -0600 (Mon, 14 Jan 2013) | 15 lines Fix XML encoding of 'identity display' in NOTIFY messages, continued. When r378933 was merged into 1.8, it should have also escaped remote_display, since it will have the same XML encoding problem when the caller/callee roles are reversed. (closes issue ABE-2902) Reported by: Guenther Kelleter ........ Merged revisions 379001 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 379020 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ r379023 | dlee | 2013-01-14 09:58:01 -0600 (Mon, 14 Jan 2013) | 20 lines Masquerades are an insane implementation detail within Asterisk. It generates a number of useless and confusing events, and manipulates channels in a way that semantically doesn't make sense. I've given a fairly thorough review of masquerade code and its usage on the wiki at https://wiki.asterisk.org/wiki/x/IwBRAQ. While ultimately it makes the most sense to abandon masquerades altogether, it will take some time to completely irradicate. Even then, there may always be code that's not worth rewriting to get rid of the masquerade. This patch does two things to make masquerades slightly less insane: * When swapping the names of the original and clone channel, only emit a single rename event of original -> original<ZOMBIE>. The original code issued three rename events to accomplish the same end. * In addition to swapping the names of the channels, also swap their uniqueid's. This allows the 'Uniqueid' field to be used as a stable identifier for a channel from and external interface, such as AMI. Review: https://reviewboard.asterisk.org/r/2266/ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@379032 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
d3bb2506a1 |
Gently reduce masquerade insanity
Masquerades are an insane implementation detail within Asterisk. It generates a number of useless and confusing events, and manipulates channels in a way that semantically doesn't make sense. I've given a fairly thorough review of masquerade code and its usage on the wiki at https://wiki.asterisk.org/wiki/x/IwBRAQ. While ultimately it makes the most sense to abandon masquerades altogether, it will take some time to completely irradicate. Even then, there may always be code that's not worth rewriting to get rid of the masquerade. This patch does two things to make masquerades slightly less insane: * When swapping the names of the original and clone channel, only emit a single rename event of original -> original<ZOMBIE>. The original code issued three rename events to accomplish the same end. * In addition to swapping the names of the channels, also swap their uniqueid's. This allows the 'Uniqueid' field to be used as a stable identifier for a channel from and external interface, such as AMI. Review: https://reviewboard.asterisk.org/r/2266/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379023 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
a91a289154 |
Fix XML encoding of 'identity display' in NOTIFY messages, continued.
When r378933 was merged into 1.8, it should have also escaped remote_display, since it will have the same XML encoding problem when the caller/callee roles are reversed. (closes issue ABE-2902) Reported by: Guenther Kelleter ........ Merged revisions 379001 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 379020 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379021 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
9e3bf9ba54 |
Merged revisions 378985 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk ................ r378985 | mjordan | 2013-01-13 16:07:00 -0600 (Sun, 13 Jan 2013) | 20 lines Reset RTP timestamp; sequence number on SSRC change In r370252 for ASTERISK-18404, Asterisk's handling of RTP was modified to better account for out of order RTP packets. This was accomplished by using the RTP timestamp and sequence number to check for out of order packets. However, when a SSRC change occurs, the timestamp and sequence number will no longer have any relation to the previously received packets. The variables tracking the timestamp and sequence number therefore have to be reset. (closes issue ASTERISK-20906) Reported by: Eelco Brolman patches: dtmf_on_hold.patch uploaded by Eelco Brolman (license #6442) ........ Merged revisions 378967 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378984 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378993 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
9b475cd3ef |
Reset RTP timestamp; sequence number on SSRC change
In r370252 for ASTERISK-18404, Asterisk's handling of RTP was modified to better account for out of order RTP packets. This was accomplished by using the RTP timestamp and sequence number to check for out of order packets. However, when a SSRC change occurs, the timestamp and sequence number will no longer have any relation to the previously received packets. The variables tracking the timestamp and sequence number therefore have to be reset. (closes issue ASTERISK-20906) Reported by: Eelco Brolman patches: dtmf_on_hold.patch uploaded by Eelco Brolman (license #6442) ........ Merged revisions 378967 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378984 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378985 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
f7f7850f7a |
Merged revisions 378935 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk ................ r378935 | dlee | 2013-01-12 00:43:37 -0600 (Sat, 12 Jan 2013) | 41 lines Fix XML encoding of 'identity display' in NOTIFY messages. XML encoding in chan_sip is accomplished by naively building the XML directly from strings. While this usually works, it fails to take into account escaping the reserved characters in XML. This patch adds an 'ast_xml_escape' function, which works similarly to 'ast_uri_encode'. This is used to properly escape the local_display attribute in XML formatted NOTIFY messages. Several things to note: * The Right Thing(TM) to do would probably be to replace the ast_build_string stuff with building an ast_xml_doc. That's a much bigger change, and out of scope for the original ticket, so I refrained myself. * It is with great sadness that I wrote my own ast_xml_escape function. There's one in libxml2, but it's knee-deep in libxml2-ness, and not easily used to one-off escape a string. * I only escaped the string we know is causing problems (local_display). At least some of the other strings are URI-encoded, which should be XML safe. Rather than figuring out what's safe and escaping what's not, it would be much cleaner to simply build an ast_xml_doc for the messages and let the XML library do the XML escaping. Like I said, that's out of scope. (closes issue ABE-2902) Reported by: Guenther Kelleter Tested by: Guenther Kelleter Review: http://reviewboard.digium.internal/r/365/ ........ Merged revision 378919 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ........ Merged revisions 378933 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378934 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378946 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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aecd2429bd |
Fix XML encoding of 'identity display' in NOTIFY messages.
XML encoding in chan_sip is accomplished by naively building the XML directly from strings. While this usually works, it fails to take into account escaping the reserved characters in XML. This patch adds an 'ast_xml_escape' function, which works similarly to 'ast_uri_encode'. This is used to properly escape the local_display attribute in XML formatted NOTIFY messages. Several things to note: * The Right Thing(TM) to do would probably be to replace the ast_build_string stuff with building an ast_xml_doc. That's a much bigger change, and out of scope for the original ticket, so I refrained myself. * It is with great sadness that I wrote my own ast_xml_escape function. There's one in libxml2, but it's knee-deep in libxml2-ness, and not easily used to one-off escape a string. * I only escaped the string we know is causing problems (local_display). At least some of the other strings are URI-encoded, which should be XML safe. Rather than figuring out what's safe and escaping what's not, it would be much cleaner to simply build an ast_xml_doc for the messages and let the XML library do the XML escaping. Like I said, that's out of scope. (closes issue ABE-2902) Reported by: Guenther Kelleter Tested by: Guenther Kelleter Review: http://reviewboard.digium.internal/r/365/ ........ Merged revision 378919 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ........ Merged revisions 378933 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378934 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378935 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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2c1720b4f7 |
Merged revisions 378915,378918 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk ................ r378915 | dlee | 2013-01-11 16:31:42 -0600 (Fri, 11 Jan 2013) | 21 lines Add JSON API for Asterisk. This provides a JSON API by pulling in and wrapping the Jansson JSON library[1]. The Asterisk API basically mirrors the Jansson functionality, with a few minor tweaks. * Some names have been asteriskified to protect the innocent. * Jansson provides both reference-stealing and reference-borrowing versions of several API's. The Asterisk API is exclusively reference-stealing for operations that put elements into arrays and objects. * No support for doubles, since we usually don't need that. * Coming along for the ride is the ast_test_validate macro, which made the unit tests much easier to write. [1]: http://www.digip.org/jansson/ (issue ASTERISK-20887) (closes issue ASTERISK-20888) Review: https://reviewboard.asterisk.org/r/2264/ ................ r378918 | file | 2013-01-11 17:05:38 -0600 (Fri, 11 Jan 2013) | 11 lines Retain XMPP filters across reconnections so external modules continue to function as expected. Previously if an XMPP client reconnected any filters added by an external module were lost. This issue exhibited itself with chan_motif not receiving and reacting to Jingle signaling. (closes issue ASTERISK-20916) Reported by: kuj ........ Merged revisions 378917 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378927 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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c5ec471766 |
Retain XMPP filters across reconnections so external modules continue to function as expected.
Previously if an XMPP client reconnected any filters added by an external module were lost. This issue exhibited itself with chan_motif not receiving and reacting to Jingle signaling. (closes issue ASTERISK-20916) Reported by: kuj ........ Merged revisions 378917 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378918 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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7695ea2643 |
Add JSON API for Asterisk.
This provides a JSON API by pulling in and wrapping the Jansson JSON library[1]. The Asterisk API basically mirrors the Jansson functionality, with a few minor tweaks. * Some names have been asteriskified to protect the innocent. * Jansson provides both reference-stealing and reference-borrowing versions of several API's. The Asterisk API is exclusively reference-stealing for operations that put elements into arrays and objects. * No support for doubles, since we usually don't need that. * Coming along for the ride is the ast_test_validate macro, which made the unit tests much easier to write. [1]: http://www.digip.org/jansson/ (issue ASTERISK-20887) (closes issue ASTERISK-20888) Review: https://reviewboard.asterisk.org/r/2264/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378915 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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919f772f78 |
Merged revisions 378889 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk ........ r378889 | rmudgett | 2013-01-09 20:40:50 -0600 (Wed, 09 Jan 2013) | 8 lines * Simplify native bridge code in ast_channel_bridge(). * Fix an unbalanced manager_bridge_event(unlink) call if AST_SOFTHANGUP_UNBRIDGE is set in ast_channel_bridge(). * Make ast_channel_bridge() use common cleanup code when leaving the bridge. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378898 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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2db3cc2e26 |
* Simplify native bridge code in ast_channel_bridge().
* Fix an unbalanced manager_bridge_event(unlink) call if AST_SOFTHANGUP_UNBRIDGE is set in ast_channel_bridge(). * Make ast_channel_bridge() use common cleanup code when leaving the bridge. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378889 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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f7e1073af5 |
Merged revisions 378874 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk ........ r378874 | rmudgett | 2013-01-09 19:43:27 -0600 (Wed, 09 Jan 2013) | 4 lines * Removed some noop code and restructured an else-if ladder in ast_generic_bridge(). * Trivial changes in ast_channel_bridge(). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378883 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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e31002b9b2 |
* Removed some noop code and restructured an else-if ladder in ast_generic_bridge().
* Trivial changes in ast_channel_bridge(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378874 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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41eba63840 |
Merged revisions 378854,378858-378859 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk ........ r378854 | rmudgett | 2013-01-09 17:22:00 -0600 (Wed, 09 Jan 2013) | 1 line Fix logger.c function definition. ........ r378858 | rmudgett | 2013-01-09 17:23:41 -0600 (Wed, 09 Jan 2013) | 6 lines Trivial misc bridge code changes. * softmix_bridge_thread() was redundantly initializing an 8K buffer. * Promoted a debug message to a warning in multiplexed_add_or_remove(). ........ r378859 | rmudgett | 2013-01-09 17:51:45 -0600 (Wed, 09 Jan 2013) | 6 lines * Simple optimization of bridge_playfile(). * Squeezed some redundancy out of update_bridge_vars(). * Wrapped long line in __ast_change_name_nolink(). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378868 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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8760b2644e |
* Simple optimization of bridge_playfile().
* Squeezed some redundancy out of update_bridge_vars(). * Wrapped long line in __ast_change_name_nolink(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378859 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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240fab21f9 |
Trivial misc bridge code changes.
* softmix_bridge_thread() was redundantly initializing an 8K buffer. * Promoted a debug message to a warning in multiplexed_add_or_remove(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378858 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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63bd8de579 |
Fix logger.c function definition.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378854 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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e844969fba |
Merged revisions 378840 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk ........ r378840 | rmudgett | 2013-01-09 16:56:08 -0600 (Wed, 09 Jan 2013) | 2 lines Trivial misc bridge code changes. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378851 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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e342ffb969 |
Trivial misc bridge code changes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378840 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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7a06fdad89 |
Merged revisions 378823 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk ........ r378823 | rmudgett | 2013-01-09 16:15:41 -0600 (Wed, 09 Jan 2013) | 2 lines Tweaked __ast_test_suite_assert_notify() and __ast_test_suite_event_notify() to be void functions. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378833 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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32cb4ca542 |
Tweaked __ast_test_suite_assert_notify() and __ast_test_suite_event_notify() to be void functions.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378823 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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613b972cc5 |
Merged revisions 378783,378789-378790 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk ................ r378783 | dlee | 2013-01-09 14:30:33 -0600 (Wed, 09 Jan 2013) | 14 lines Fix end condition in ast_rtp_lookup_mime_multiple2. The erroneous end condition would never include the AST_RTP_CISCO_DTMF flag in the debug output. (closes issue ASTERISK-20772) Reported by: Xavier Hienne ........ Merged revisions 378776 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378780 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ r378789 | rmudgett | 2013-01-09 14:56:23 -0600 (Wed, 09 Jan 2013) | 4 lines * Found some more places to use ast_channel_lock_both(). * Minor optimization in ast_rtp_instance_early_bridge(). ................ r378790 | rmudgett | 2013-01-09 15:14:39 -0600 (Wed, 09 Jan 2013) | 4 lines * Whitespace changes. * Made ast_test_init() match its prototype. ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378801 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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04501f0054 |
* Whitespace changes.
* Made ast_test_init() match its prototype. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378790 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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671499c8b2 |
* Found some more places to use ast_channel_lock_both().
* Minor optimization in ast_rtp_instance_early_bridge(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378789 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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6dfcc86c0d |
Fix end condition in ast_rtp_lookup_mime_multiple2.
The erroneous end condition would never include the AST_RTP_CISCO_DTMF flag in the debug output. (closes issue ASTERISK-20772) Reported by: Xavier Hienne ........ Merged revisions 378776 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378780 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378783 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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2ba33b4c3a |
Merged revisions 378735,378748 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk ................ r378735 | dlee | 2013-01-09 13:38:53 -0600 (Wed, 09 Jan 2013) | 13 lines Replace errant tabs with spaces in causes.h. (closes issue ASTERISK-20826) Reported by: snuffy Patches: notabs.dif uploaded by snuffy (license 5024) ........ Merged revisions 378733 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378734 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ r378748 | dlee | 2013-01-09 14:12:00 -0600 (Wed, 09 Jan 2013) | 13 lines Move declaration of ast_regex_string_to_regex_pattern futher down strings.h. The prior location is before the declaration of struct ast_str, which causes compiler warnings. (closes issue ASTERISK-20852) Reported by: Pavel Troller Patches: strings.diff uploaded by Pavel Troller (license 6302) ........ Merged revisions 378747 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378759 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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2f25e4b83e |
Move declaration of ast_regex_string_to_regex_pattern futher down strings.h.
The prior location is before the declaration of struct ast_str, which causes compiler warnings. (closes issue ASTERISK-20852) Reported by: Pavel Troller Patches: strings.diff uploaded by Pavel Troller (license 6302) ........ Merged revisions 378747 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378748 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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4f4d6da941 |
Replace errant tabs with spaces in causes.h.
(closes issue ASTERISK-20826) Reported by: snuffy Patches: notabs.dif uploaded by snuffy (license 5024) ........ Merged revisions 378733 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378734 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378735 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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2fa8cc3f51 |
Merged revisions 378688,378691 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk ................ r378688 | rmudgett | 2013-01-08 17:44:26 -0600 (Tue, 08 Jan 2013) | 35 lines app_queue: Fix multiple calls to a queue member that is in only one queue. When ringinuse=no queue members can receive more than one call if these calls happen at nearly the same time. * Fix so a queue member does not receive more than one call from a queue. NOTE: This fix does not prevent multiple calls to a member if the member is in more than one queue. * Did some refactoring to eliminate some code redundancy. (issue ASTERISK-16115) Reported by: nik600 Patches: jira_asterisk_16115_single_q_v1.8.patch (license #5621) patch uploaded by rmudgett Modified * Revert the -r341580 and -r341599 changes adding the queues.conf check_state_unknown option as it was added in an attempt to fix this problem. The fix did not need to be optional. The fix should not have tried to explicitly set the device state. Setting the device state by something other than the device introduces a race condition. I also could not see how the change would be effective other than delaying the app_queue code long enough for the device state to propagate to app_queue. ........ Merged revisions 378663 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378683 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 378687 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ r378691 | rmudgett | 2013-01-08 18:05:35 -0600 (Tue, 08 Jan 2013) | 10 lines app_queue: Fix incorrect assertion. (issue ASTERISK-16115) ........ Merged revisions 378689 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 378690 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378699 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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c41e70d647 |
app_queue: Fix incorrect assertion.
(issue ASTERISK-16115) ........ Merged revisions 378689 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 378690 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378691 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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8ed2c74fe3 |
app_queue: Fix multiple calls to a queue member that is in only one queue.
When ringinuse=no queue members can receive more than one call if these calls happen at nearly the same time. * Fix so a queue member does not receive more than one call from a queue. NOTE: This fix does not prevent multiple calls to a member if the member is in more than one queue. * Did some refactoring to eliminate some code redundancy. (issue ASTERISK-16115) Reported by: nik600 Patches: jira_asterisk_16115_single_q_v1.8.patch (license #5621) patch uploaded by rmudgett Modified * Revert the -r341580 and -r341599 changes adding the queues.conf check_state_unknown option as it was added in an attempt to fix this problem. The fix did not need to be optional. The fix should not have tried to explicitly set the device state. Setting the device state by something other than the device introduces a race condition. I also could not see how the change would be effective other than delaying the app_queue code long enough for the device state to propagate to app_queue. ........ Merged revisions 378663 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378683 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 378687 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378688 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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99701cd1d1 |
Remove tasks from the taskprocessor and free them when taskprocessor is destroyed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378654 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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a08847c270 |
Add some doxygen and remove an unnecessary unlock.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378653 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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bdd8da406b |
Address review board feedback from Matt and Richard
* Remove extraneous whitespace * Bump up debug levels of messages and add identifying info to messages. * Account for potential failures of ao2_link() * Add additional test and some more test data * Add some comments in places where they could be useful * Make threadpool listeners and their callbacks optional git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378652 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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ad73fe2c9f |
Merged revisions 378634 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk ........ r378634 | wedhorn | 2013-01-06 15:37:59 -0600 (Sun, 06 Jan 2013) | 6 lines Skinny blob cleanup Cleanup of red blobs in chan_skinny and possible other small formatting issues. Review: https://reviewboard.asterisk.org/r/2262/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378639 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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7d5345c9c0 |
Skinny blob cleanup
Cleanup of red blobs in chan_skinny and possible other small formatting issues. Review: https://reviewboard.asterisk.org/r/2262/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378634 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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00b40f6e2a |
Merged revisions 378623-378624 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk ................ r378623 | wedhorn | 2013-01-06 14:45:12 -0600 (Sun, 06 Jan 2013) | 12 lines Rewrite skinny dialing to remove threaded simpleswitch This rewrite changes skinny dialing from the threaded simpleswitch to a scheduled timeout approach. There were some underlying issues with the threaded simple switch with occasional corruption and possible segfaults. Review: https://reviewboard.asterisk.org/r/2240/ ........ Merged revisions 378622 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ r378624 | wedhorn | 2013-01-06 15:09:43 -0600 (Sun, 06 Jan 2013) | 6 lines Add group and namedgroup pickup to skinny Above says it all. Code by snuff, cleaned up by me. Review: https://reviewboard.asterisk.org/r/2246/ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378629 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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f795062662 |
Add group and namedgroup pickup to skinny
Above says it all. Code by snuff, cleaned up by me. Review: https://reviewboard.asterisk.org/r/2246/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378624 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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bacc5e6604 |
Rewrite skinny dialing to remove threaded simpleswitch
This rewrite changes skinny dialing from the threaded simpleswitch to a scheduled timeout approach. There were some underlying issues with the threaded simple switch with occasional corruption and possible segfaults. Review: https://reviewboard.asterisk.org/r/2240/ ........ Merged revisions 378622 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378623 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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2765de67b6 |
Merged revisions 378593 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk ................ r378593 | jrose | 2013-01-04 17:14:54 -0600 (Fri, 04 Jan 2013) | 23 lines res_srtp: Prevent a crash from occurring due to srtp_create failures in srtp_create Under some circumstances, libsrtp's srtp_create function deallocates memory that it wasn't initially responsible for allocating. Because we weren't initially aware of this behavior, this memory was still used in spite of being unallocated during the course of the srtp_unprotect function. A while back I made a patch which would set this value to NULL, but that exposed a possible condition where we would then try to check a member of the struct which would cause a segfault. In order to address these problems, ast_srtp_unprotect will now set an error value when it ends without a valid SRTP session which will result in the caller of srtp_unprotect observing this error and hanging up the relevant channel instead of trying to keep using the invalid session address. (closes issue ASTERISK-20499) Reported by: Tootai Review: https://reviewboard.asterisk.org/r/2228/diff/#index_header ........ Merged revisions 378591 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378592 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378600 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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9d5f6e050e |
res_srtp: Prevent a crash from occurring due to srtp_create failures in srtp_create
Under some circumstances, libsrtp's srtp_create function deallocates memory that it wasn't initially responsible for allocating. Because we weren't initially aware of this behavior, this memory was still used in spite of being unallocated during the course of the srtp_unprotect function. A while back I made a patch which would set this value to NULL, but that exposed a possible condition where we would then try to check a member of the struct which would cause a segfault. In order to address these problems, ast_srtp_unprotect will now set an error value when it ends without a valid SRTP session which will result in the caller of srtp_unprotect observing this error and hanging up the relevant channel instead of trying to keep using the invalid session address. (closes issue ASTERISK-20499) Reported by: Tootai Review: https://reviewboard.asterisk.org/r/2228/diff/#index_header ........ Merged revisions 378591 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378592 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378593 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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77150eecd9 |
Merged revisions 378565,378585 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk ................ r378565 | elguero | 2013-01-04 15:20:12 -0600 (Fri, 04 Jan 2013) | 27 lines Fix SIP Notify Messages To Have The Proper IP Address In The FROM Field On a multihomed server when sending a NOTIFY message, we were not figuring out which network should be used to contact the peer. This patch fixes the problem by calling ast_sip_ouraddrfor() and then build_via() so that our NOTIFY message contains the correct IP address. Also, a debug message is being added to help follow the call-id changes that occur. This was helpful for confirming that the IP address was set properly since the call-id contains the IP address. It also will be helpful for troubleshooting purposes when following a call in the debug logs. (closes issue ASTERISK-20805) Reported by: Bryan Hunt Tested by: Bryan Hunt, Michael L. Young Patches: asterisk-20805-notify-ip-v2.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2255/ ........ Merged revisions 378554 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378559 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ r378585 | kmoore | 2013-01-04 16:19:16 -0600 (Fri, 04 Jan 2013) | 13 lines Fix pjproject compilation in certain circumstances On a fresh checkout of Asterisk 11, running make before ./configure could cause the pjproject subdirectory to get in an odd state that would prevent compilation. This patch by Tilghman prevents that from occurring. (closes issue ASTERISK-20681) Patch-by: Tilghman Lesher ........ Merged revisions 378582 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378586 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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9e814816cb |
Fix pjproject compilation in certain circumstances
On a fresh checkout of Asterisk 11, running make before ./configure could cause the pjproject subdirectory to get in an odd state that would prevent compilation. This patch by Tilghman prevents that from occurring. (closes issue ASTERISK-20681) Reported by: Dinesh Ramjuttun Tested by: danilo borges, Steve Lang patches: 20121208__ccar_solved.diff.txt uploaded by Tilghman Lesher (license 5003) ........ Merged revisions 378582 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378585 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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209373262d |
Fix SIP Notify Messages To Have The Proper IP Address In The FROM Field
On a multihomed server when sending a NOTIFY message, we were not figuring out which network should be used to contact the peer. This patch fixes the problem by calling ast_sip_ouraddrfor() and then build_via() so that our NOTIFY message contains the correct IP address. Also, a debug message is being added to help follow the call-id changes that occur. This was helpful for confirming that the IP address was set properly since the call-id contains the IP address. It also will be helpful for troubleshooting purposes when following a call in the debug logs. (closes issue ASTERISK-20805) Reported by: Bryan Hunt Tested by: Bryan Hunt, Michael L. Young Patches: asterisk-20805-notify-ip-v2.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2255/ ........ Merged revisions 378554 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378559 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378565 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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98fe918625 |
Merged revisions 378557 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk ................ r378557 | file | 2013-01-04 15:18:07 -0600 (Fri, 04 Jan 2013) | 11 lines Don't pass STUN packets through the SRTP unprotect function. (closes issue AST-1036) Reported by: jbigelow ........ Merged revisions 378553 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378555 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378560 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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4838d6ff68 |
Don't pass STUN packets through the SRTP unprotect function.
(closes issue AST-1036) Reported by: jbigelow ........ Merged revisions 378553 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378555 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378557 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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cd0aef75ca |
Merged revisions 378543 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk ........ r378543 | lathama | 2013-01-04 10:44:33 -0600 (Fri, 04 Jan 2013) | 6 lines Doxygen Cleanups Baseline clean up of formating to make room for extended documentation (issue ASTERISK-20259) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378548 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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a30e74da4f |
Doxygen Cleanups
Baseline clean up of formating to make room for extended documentation (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378543 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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7980f15ada |
Merged revisions 378516 via svnmerge from
file:///srv/subversion/repos/asterisk/trunk ................ r378516 | elguero | 2013-01-03 16:14:20 -0600 (Thu, 03 Jan 2013) | 25 lines Fix Queue Log Reporting Every Call COMPLETECALLER With "h" Extension Present When the "h" extension is present within the context of the queue, all calls are being reported COMPLETECALLER even when the agent is hanging up the call. This patch checks to see if the agent hung-up or not instead of only relying on checking if the queue (caller) channel hung-up or not. It would appear that having the h extension in the mix, the pbx goes to the h extension, "hanging-up" the queue channel and triggering the reporting of COMPLETECALLER. (closes issue ASTERISK-20743) Reported by: call Tested by: call, Michael L. Young Patches: asterisk-20743-q-cmplt-caller.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2256/ ........ Merged revisions 378514 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378515 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378523 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |