Again, since res_jabber/res_xmpp have duplicate APIs, their documentation ref
links have to specify which reference they're referring to. The various
documentation parsers can interpret the module attribute however they want
in order to construct the appropriate links.
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r379211 | mjordan | 2013-01-16 09:33:05 -0600 (Wed, 16 Jan 2013) | 21 lines
Multiple revisions 379209-379210
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r379209 | mjordan | 2013-01-16 09:27:44 -0600 (Wed, 16 Jan 2013) | 8 lines
Add module tags to documentation for res_jabber/res_xmpp
Since res_jabber/res_xmpp provide the same APIs (app/func/manager/etc.),
the XML documentation for each needs to call out which module is providing
the documentation. The module attribute has been added to the various XML
fragments for this purpose.
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r379210 | mjordan | 2013-01-16 09:30:20 -0600 (Wed, 16 Jan 2013) | 4 lines
Update the dtd to actually *support* the module attribute in all elements
Mea culpa.
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r379209 | mjordan | 2013-01-16 09:27:44 -0600 (Wed, 16 Jan 2013) | 8 lines
Add module tags to documentation for res_jabber/res_xmpp
Since res_jabber/res_xmpp provide the same APIs (app/func/manager/etc.),
the XML documentation for each needs to call out which module is providing
the documentation. The module attribute has been added to the various XML
fragments for this purpose.
........
r379210 | mjordan | 2013-01-16 09:30:20 -0600 (Wed, 16 Jan 2013) | 4 lines
Update the dtd to actually *support* the module attribute in all elements
Mea culpa.
........
Merged revisions 379209-379210 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379211 65c4cc65-6c06-0410-ace0-fbb531ad65f3
file:///srv/subversion/repos/asterisk/trunk
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r379180 | mjordan | 2013-01-15 22:14:38 -0600 (Tue, 15 Jan 2013) | 27 lines
Fix parsing SMSSRC for SMS messages
The parser for SMS messages would incorrectly parse out the from number.
The parsing would incorrectly start scanning for the from number at the
same index as the first double quote ("); this would inadvertently cause
it to treat the first double quote as the terminating double quote for
the from number as well.
The SMSSRC should now populate correctly.
(closes issue ASTERISK-16822)
Reported by: menschentier
Tested by: Jonas Falck
patches:
fixSMSSRC.patch uploaded by jonax (license 6320)
(closes issue ASTERISK-19153)
Reported by: Panos Gkikakis
patches:
sms-sender-fix.diff uploaded by roeften (license 5884)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@379189 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The parser for SMS messages would incorrectly parse out the from number.
The parsing would incorrectly start scanning for the from number at the
same index as the first double quote ("); this would inadvertently cause
it to treat the first double quote as the terminating double quote for
the from number as well.
The SMSSRC should now populate correctly.
(closes issue ASTERISK-16822)
Reported by: menschentier
Tested by: Jonas Falck
patches:
fixSMSSRC.patch uploaded by jonax (license 6320)
(closes issue ASTERISK-19153)
Reported by: Panos Gkikakis
patches:
sms-sender-fix.diff uploaded by roeften (license 5884)
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Merged revisions 379178 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r379144 | mjordan | 2013-01-15 17:54:34 -0600 (Tue, 15 Jan 2013) | 17 lines
Add busy detection to chan_mobile
From the patch author:
"First this patch adds general support for busy detection. It also adds support
for the ECAM command at Sony Ericsson phones and also signals busy when only
early media was received but the call got not answered."
Review: https://reviewboard.asterisk.org/r/323
(closes issue ASTERISK-14527)
Reported by: Artem Makhutov
Tested by: Artem Makhutov
patches:
busy-full5.patch uploaded by artem (license 5757)
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r379147 | mjordan | 2013-01-15 18:16:22 -0600 (Tue, 15 Jan 2013) | 25 lines
Set the INVALID_EXTEN channel variable when chan_misdn forces the 'i' extension
The chan_misdn channel driver will send a channel with an invalid destination
to the 'i' extension itself if said extension can be reached. It forgot,
however, to set the INVALID_EXTEN channel variable when it bounces the channel
to this extension. Dialplan writers everywhere moaned at yet another
inconsistency.
This is yet another example of why duplicating logic in multiple places results
in bugs that stick around in Jira for just under three years.
Yes: ASTERISK-15456 was created on January 18th, 2010. Patch committed on
January 15th, 2013. Ouch.
(closes issue ASTERISK-15456)
Reported by: Thomas Omerzu
patches:
chan_misdn_invalid.patch2 uploaded by Thomas Omerzu (license 5927)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@379156 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The chan_misdn channel driver will send a channel with an invalid destination
to the 'i' extension itself if said extension can be reached. It forgot,
however, to set the INVALID_EXTEN channel variable when it bounces the channel
to this extension. Dialplan writers everywhere moaned at yet another
inconsistency.
This is yet another example of why duplicating logic in multiple places results
in bugs that stick around in Jira for just under three years.
Yes: ASTERISK-15456 was created on January 18th, 2010. Patch committed on
January 15th, 2013. Ouch.
(closes issue ASTERISK-15456)
Reported by: Thomas Omerzu
patches:
chan_misdn_invalid.patch2 uploaded by Thomas Omerzu (license 5927)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379147 65c4cc65-6c06-0410-ace0-fbb531ad65f3
From the patch author:
"First this patch adds general support for busy detection. It also adds support
for the ECAM command at Sony Ericsson phones and also signals busy when only
early media was received but the call got not answered."
Review: https://reviewboard.asterisk.org/r/323
(closes issue ASTERISK-14527)
Reported by: Artem Makhutov
Tested by: Artem Makhutov
patches:
busy-full5.patch uploaded by artem (license 5757)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379144 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Clarify some documentation
* Change copyright date of taskprocessor files
* Address potential issue of creating taskprocessor with listener if
taskprocessor with that name exists already
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@379124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Now user data is allocated by the creator of the taskprocessor
listener and that user data is passed into ast_taskprocessor_listener_alloc().
Similarly, freeing of the user data is left up to the user himself. He can
free the data when the taskprocessor shuts down, or he can choose to hold
onto it if it makes sense to do so.
This, unsurprisingly, makes threadpool allocation a LOT cleaner now.
git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@379120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
file:///srv/subversion/repos/asterisk/trunk
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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
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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/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@379032 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
file:///srv/subversion/repos/asterisk/trunk
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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)
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Merged revisions 378967 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378993 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
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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/
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Merged revisions 378933 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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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/
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Merged revisions 378933 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 378934 from http://svn.asterisk.org/svn/asterisk/branches/11
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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/
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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
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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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378918 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
file:///srv/subversion/repos/asterisk/trunk
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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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378898 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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
* 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
* 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
file:///srv/subversion/repos/asterisk/trunk
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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
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Merged revisions 378776 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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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().
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r378790 | rmudgett | 2013-01-09 15:14:39 -0600 (Wed, 09 Jan 2013) | 4 lines
* Whitespace changes.
* Made ast_test_init() match its prototype.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@378801 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
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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.
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r378691 | rmudgett | 2013-01-08 18:05:35 -0600 (Tue, 08 Jan 2013) | 10 lines
app_queue: Fix incorrect assertion.
(issue ASTERISK-16115)
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