[asterisk-commits] may: branch may/ooh323_qsig r369549 - in /team/may/ooh323_qsig: ./ addons/ ad...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Jul 3 05:52:51 CDT 2012
Author: may
Date: Tue Jul 3 05:52:25 2012
New Revision: 369549
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=369549
Log:
Multiple revisions 369013,369027-369028,369034,369057,369061,369068,369086,369092,369110,369126,369141-369142,369148,369166-369167,369184-369185,369190,369207,369216-369217,369237,369240,369260-369261,369264,369295-369296,369304,369326,369329,369346,369367,369370,369386,369392,369414,369449,369454,369473,369489,369492-369493
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r369013 | kpfleming | 2012-06-15 20:20:16 +0400 (Fri, 15 Jun 2012) | 25 lines
Multiple revisions 369001-369002
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r369001 | kpfleming | 2012-06-15 10:56:08 -0500 (Fri, 15 Jun 2012) | 11 lines
Add support-level indications to many more source files.
Since we now have tools that scan through the source tree looking for files
with specific support levels, we need to ensure that every file that is
a component of a 'core' or 'extended' module (or the main Asterisk binary)
is explicitly marked with its support level. This patch adds support-level
indications to many more source files in tree, but avoids adding them to
third-party libraries that are included in the tree and to source files
that don't end up involved in Asterisk itself.
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r369002 | kpfleming | 2012-06-15 10:57:14 -0500 (Fri, 15 Jun 2012) | 3 lines
Add a script to enable finding source files without support-levels defined.
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Merged revisions 369001-369002 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 369005 from http://svn.asterisk.org/svn/asterisk/branches/10
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r369027 | qwell | 2012-06-15 20:30:58 +0400 (Fri, 15 Jun 2012) | 8 lines
Fix voicemail API tests by using the correct argument order for create/destroy.
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Merged revisions 369024 from http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11
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Merged revisions 369026 from http://svn.asterisk.org/svn/asterisk/branches/10-digiumphones
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r369028 | kmoore | 2012-06-15 21:13:20 +0400 (Fri, 15 Jun 2012) | 16 lines
Allow chan_sip to decline unwanted media streams
This change replaces the static array of four representable media
streams with an AST_LIST so that chan_sip can keep track of offered
media streams. This allows chan_sip to deal with offers containing
multiple same-type streams and many other situations without rejecting
the SDP offer in its entirety, yet still generating a valid response.
This also covers cases where Asterisk can not comprehend the offer if
it is in the correct format.
Previously, chan_sip would reject SDP offers or entirely ignore
individual stream offers in an effort to be more compatible which
would often result in invalid SDP responses.
Review: https://reviewboard.asterisk.org/r/1988/
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r369034 | wedhorn | 2012-06-16 01:18:56 +0400 (Sat, 16 Jun 2012) | 14 lines
Various small chan_skinny fixes and cleanup
Added test to skinny_register to only allow device to register against
a device that is not already registered.
Addback l->device test for skinny_show_lines. Fixes segfault if a line
is configured but not configured to a device. Reverses part of r368680.
Removed redundant l->device tests in subsubstate and dumpsub. l->device
will always be valid if these routines are called. Reverses 368948 -
discussed with mjordan on irc.
Some indentation cleanup.
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r369057 | rmudgett | 2012-06-18 22:25:22 +0400 (Mon, 18 Jun 2012) | 15 lines
Fix monitoring calls put in a parking lot.
* Fix a regression that was introduced by -r366167 which effectively
disabled monitoring parked calls.
(closes issue ASTERISK-20012)
Reported by: sdolloff
Tested by: rmudgett
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Merged revisions 369043 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 369044 from http://svn.asterisk.org/svn/asterisk/branches/10
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r369061 | kmoore | 2012-06-19 02:56:01 +0400 (Tue, 19 Jun 2012) | 8 lines
Fix AST_CONTROL_PVT_CAUSE_CODE handling
When the IAX2 Who Hung Up? changes were added, they uncovered a bug in
the way AST_CONTROL_PVT_CAUSE_CODE was handled in
feature_request_and_dial(). This particular frame subtype was being
treated like more terminal control frames causing the function to be
exited prematurely.
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r369068 | mmichelson | 2012-06-19 19:44:42 +0400 (Tue, 19 Jun 2012) | 20 lines
Fix request routing issue when outboundproxy is used.
Asterisk was incorrectly setting the destination of CANCELs
and ACKs for error responses to the URI of the initial INVITE.
This resulted in further requests, such as INVITEs with authentication
credentials, to be routed incorrectly. Instead, when these CANCEL
or ACKs are to be sent, we should simply keep the destination the
same as what it previously was. There is no need to alter it any.
(closes issue ASTERISK-20008)
Reported by Marcus Hunger
Patches:
ASTERISK-20008.patch uploaded by Mark Michelson (license #5049)
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Merged revisions 369066 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 369067 from http://svn.asterisk.org/svn/asterisk/branches/10
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r369086 | kmoore | 2012-06-20 01:13:41 +0400 (Wed, 20 Jun 2012) | 10 lines
Ensure that pvt cause information does not break native bridging
Channel drivers that allow native bridging need to handle
AST_CONTROL_PVT_CAUSE_CODE frames and previously did not handle them
properly, usually breaking out of the native bridge. This change
corrects that behavior and exposes the available cause code information
to the dialplan while native bridges are in place. This required
exposing the HANGUPCAUSE hash setter outside of channel.c, so
additional documentation has been added.
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r369092 | may | 2012-06-20 03:36:43 +0400 (Wed, 20 Jun 2012) | 11 lines
check rtptimeouts in ooh323 channels as per config file
(rtp voice, video, udptl except rtcp)
(closes issue ASTERISK-19179)
Reported by: TSAREGORODTSEV Yury
Patches:
19179-ooh323-ast10.patch
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Merged revisions 369091 from http://svn.asterisk.org/svn/asterisk/branches/10
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r369110 | elguero | 2012-06-20 06:07:00 +0400 (Wed, 20 Jun 2012) | 26 lines
Fix NULL pointer segfault in ast_sockaddr_parse()
While working with ast_parse_arg() to perform a validity check, a segfault
occurred. The segfault occurred due to passing a NULL pointer to
ast_sockaddr_parse() from ast_parse_arg(). According to the documentation in
config.h, "result pointer to the result. NULL is valid here, and can be used to
perform only the validity checks."
This patch fixes the segfault by checking for a NULL pointer. This patch also
adds documentation to netsock2.h about why it is necessary to check for a NULL
pointer.
(Closes issue ASTERISK-20006)
Reported by: Michael L. Young
Tested by: Michael L. Young
Patches:
asterisk-20006-netsock-null-ptr.diff uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/1990/
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Merged revisions 369108 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 369109 from http://svn.asterisk.org/svn/asterisk/branches/10
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r369126 | elguero | 2012-06-20 07:18:50 +0400 (Wed, 20 Jun 2012) | 12 lines
Add IPv6 Support To Manager
This patch adds IPv6 support to AMI.
(Closes issue ASTERISK-19965)
Reported by: Michael L. Young
Tested by: Michael L. Young
Patches:
ami_ipv6_v3.diff uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/1968/
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r369141 | may | 2012-06-20 15:20:05 +0400 (Wed, 20 Jun 2012) | 2 lines
use right definition for channel name
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r369142 | seanbright | 2012-06-20 15:47:12 +0400 (Wed, 20 Jun 2012) | 2 lines
Remove declaration of eivr_connect_socket because it no longer exists.
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r369148 | may | 2012-06-20 21:48:20 +0400 (Wed, 20 Jun 2012) | 13 lines
fix locking issue on empty callList
(issue ASTERISK-19298)
Reported by:
Dmitry Melekhov
Patches:
ASTERISK-18322-2.patch
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Merged revisions 369146 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 369147 from http://svn.asterisk.org/svn/asterisk/branches/10
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r369166 | rmudgett | 2012-06-21 01:32:40 +0400 (Thu, 21 Jun 2012) | 1 line
Fix chan_misdn compile error.
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r369167 | rmudgett | 2012-06-21 01:33:11 +0400 (Thu, 21 Jun 2012) | 5 lines
Don't waste time initializing the whole call_identifer_str[].
The array is either setup with a callid string or only the first element
needs to be initialized.
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r369184 | kmoore | 2012-06-22 18:57:07 +0400 (Fri, 22 Jun 2012) | 8 lines
Add "Who Hung Up?" implementation for DAHDI SS7 subtechnology
Testing was done on a local machine to verify that protocol and
cause information was being sent properly.
Review: https://reviewboard.asterisk.org/r/1955/
(issue SWP-4222)
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r369185 | kmoore | 2012-06-22 19:10:38 +0400 (Fri, 22 Jun 2012) | 10 lines
Add HANGUPCAUSE hash support for analog and PRI DAHDI subtechs
This is part of the DAHDI support for the Asterisk 11 "Who Hung Up?"
project and covers the implementation for the technologies implemented
in sig_analog.c and sig_pri.c. Tested on a local machine to verify
protocol and cause information is available.
Review: https://reviewboard.asterisk.org/r/1953/
(issue SWP-4222)
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r369190 | kmoore | 2012-06-22 19:57:02 +0400 (Fri, 22 Jun 2012) | 8 lines
Add HANGUPCAUSE hash implementation for DAHDI MFC/R2 subtech
This adds a minimal implementation of the "Who Hung Up?" Asterisk 11
work to chan_dahdi.c for the MFC/R2 DAHDI subtech. Given the way that
OpenR2 interfaces with chan_dahdi, it is much harder to expose the
type of protocol information that is available in PRI, SS7, or other
channel technologies.
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r369207 | kmoore | 2012-06-22 21:25:06 +0400 (Fri, 22 Jun 2012) | 14 lines
Don't parse media stream state for SIP video streams
The sendonly/recvonly/sendrecv/inactive media stream attributes were
parsed for video, but nothing was ever done with them. With this code
removed, an UNSUPPORTED message is produced when these attributes are
used in conjunction with a video stream which is the better behavior
since they were never really supported in the first place.
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Merged revisions 369195 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 369206 from http://svn.asterisk.org/svn/asterisk/branches/10
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r369216 | kmoore | 2012-06-22 23:54:41 +0400 (Fri, 22 Jun 2012) | 2 lines
Fix wrong variable name in the R2 disconnect callback
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r369217 | twilson | 2012-06-23 00:05:22 +0400 (Sat, 23 Jun 2012) | 14 lines
Don't crash on a guest directmedia call
A sip_pvt may not have relatedpeer set if a call doesn't match up
with a peer. If there is no relatedpeer, there is no direct media
ACL to apply, so just return that it is allowed.
(closes issue ASTERISK-20040)
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Merged revisions 369214 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 369215 from http://svn.asterisk.org/svn/asterisk/branches/10
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r369237 | rmudgett | 2012-06-23 00:52:54 +0400 (Sat, 23 Jun 2012) | 8 lines
Change incorrect chan_sip zombie hangup debug message. They are all zombies now.
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Merged revisions 369235 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 369236 from http://svn.asterisk.org/svn/asterisk/branches/10
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r369240 | rmudgett | 2012-06-23 01:06:36 +0400 (Sat, 23 Jun 2012) | 8 lines
Check if PBX was started for generic CCSS recall.
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Merged revisions 369238 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 369239 from http://svn.asterisk.org/svn/asterisk/branches/10
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r369260 | rmudgett | 2012-06-23 01:43:44 +0400 (Sat, 23 Jun 2012) | 8 lines
Check if PBX was started and fix F and F(x) action logic in Dial application.
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Merged revisions 369258 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 369259 from http://svn.asterisk.org/svn/asterisk/branches/10
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r369261 | rmudgett | 2012-06-23 01:51:05 +0400 (Sat, 23 Jun 2012) | 1 line
Fix F and F(x) action logic in Queue application.
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r369264 | rmudgett | 2012-06-23 02:12:06 +0400 (Sat, 23 Jun 2012) | 8 lines
Explicitly check caller hangup in app Queue rather than a polluted res2 value.
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Merged revisions 369262 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 369263 from http://svn.asterisk.org/svn/asterisk/branches/10
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r369295 | rmudgett | 2012-06-23 04:29:18 +0400 (Sat, 23 Jun 2012) | 25 lines
Fix Bridge application and AMI Bridge action error handling.
* Fix AMI Bridge action disconnecting the AMI link on error.
* Fix AMI Bridge action and Bridge application not checking if their
masquerades were successful.
* Fix Bridge application running the h-exten when it should not.
* Made do_bridge_masquerade() return if the masquerade was successful so
the Bridge application and AMI Bridge action could deal with it correctly.
* Made bridge_call_thread_launch() hangup the passed in channels if the
bridge_call_thread fails to start. Those channels would have been
orphaned.
* Made builtin_atxfer() check the success of the transfer masquerade
setup.
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Merged revisions 369282 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 369283 from http://svn.asterisk.org/svn/asterisk/branches/10
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r369296 | rmudgett | 2012-06-23 04:33:41 +0400 (Sat, 23 Jun 2012) | 1 line
Fix F and F(x) action logic in Bridge application.
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r369304 | mmichelson | 2012-06-25 18:30:19 +0400 (Mon, 25 Jun 2012) | 17 lines
Be more consistent with the return code for requests received from invalid domain.
When Asterisk receives an INVITE from an external domain when allowexternaldomains=no
send a 403 instead of a 404. This is consistent with Asterisk's behavior when receiving
a REGISTER in this situation.
(Closes issue ASTERISK-19601)
Reported by Matthew Jordan
Patches:
ASTERISK-19601-no401.patch uploaded by Mark Michelson (License #5049)
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Merged revisions 369302 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 369303 from http://svn.asterisk.org/svn/asterisk/branches/10
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r369326 | mmichelson | 2012-06-25 19:55:25 +0400 (Mon, 25 Jun 2012) | 23 lines
Multiple revisions 369323-369324
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r369323 | mmichelson | 2012-06-25 10:35:43 -0500 (Mon, 25 Jun 2012) | 9 lines
Eliminate embedding of res_adsi.so module.
The way this is done is to stop using the optional API.
Instead, res_adsi.so, when loaded fills in a table of
function pointers.
Review: https://reviewboard.asterisk.org/r/1991
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r369324 | mmichelson | 2012-06-25 10:50:17 -0500 (Mon, 25 Jun 2012) | 2 lines
Forgot to svn add this file in my last commit.
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Merged revisions 369323-369324 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 369325 from http://svn.asterisk.org/svn/asterisk/branches/10
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r369329 | rmudgett | 2012-06-25 20:07:02 +0400 (Mon, 25 Jun 2012) | 18 lines
Fix Bridge application occasionally returning to the wrong location.
* Fix do_bridge_masquerade() getting the resume location from the zombie
channel. The code must not touch a clone channel after it has masqueraded
it. The clone channel has become a zombie and is starting to hangup.
(closes issue ASTERISK-19985)
Reported by: jamicque
Patches:
jira_asterisk_19985_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: jamicque
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Merged revisions 369327 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 369328 from http://svn.asterisk.org/svn/asterisk/branches/10
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r369346 | mjordan | 2012-06-25 21:59:34 +0400 (Mon, 25 Jun 2012) | 21 lines
Add AMI event documentation
This patch adds the core changes necessary to support AMI event documentation
in the source files of Asterisk, and adds documentation to those AMI events
defined in the core application modules. Event documentation is built from
the source by two new python scripts, located in build_tools:
get_documentation.py and post_process_documentation.py.
The get_documentation.py script mirrors the actions of the existing AWK
get_documentation scripts, except that it will scan the entirety of a source
file for Asterisk documentation. Upon encountering it, if the documentation
happens to be an AMI event, it will attempt to extract information about the
event directly from the manager event macro calls that raise the event. The
post_process_documentation.py script combines manager event instances that
are the same event but documented in multiple source files. It generates
the final core-[lang].xml file.
As this process can take longer to complete than a typical 'make all', it
is only performed if a new make target, 'full', is chosen.
Review: https://reviewboard.asterisk.org/r/1967/
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r369367 | mmichelson | 2012-06-25 23:26:31 +0400 (Mon, 25 Jun 2012) | 17 lines
Re-fix how local tag is generated when sending a 481 to an INVITE.
Match our local tag to whatever to-tag was sent in the initial INVITE.
Because the size of the to-tag may not fit in the buffer in the sip_pvt,
it has been changed to a string field.
(closes issue ASTERISK-19892)
reported by Walter Doekes
Review: https://reviewboard.asterisk.org/r/1977
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Merged revisions 369352 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 369353 from http://svn.asterisk.org/svn/asterisk/branches/10
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r369370 | mjordan | 2012-06-25 23:39:03 +0400 (Mon, 25 Jun 2012) | 32 lines
Fix incorrect duration reporting in CDRs created in batch mode
Certain places in core/cdr.c would, if the duration value were 0, calculate the
duration as being the delta between the current time and the time at which the
CDR record was started. While this does not typically cause a problem in
non-batch mode, this can cause an issue in batch mode where CDR records are
gathered and written long after those calls have ended. In particular, this
affects calls that were never answered, as those are expected to have a duration
of 0. Often, this would result in CDR logs with a significant number of calls
with lengthy durations, but dispositions of "BUSY".
Note that this does not affect cdr_csv, as that backend does not use
ast_cdr_getvar and instead directly reports the duration value. The affected
core backends include cdr_apative_odbc and cdr_custom; other extended or
deprecated CDR backends may potentially still directly manipulate the duration
values.
(issue ASTERISK-19860)
Reported by: Thomas Arimont
(issue AST-883)
Reported by: Thomas Arimont
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1996/
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Merged revisions 369369 from http://svn.asterisk.org/svn/asterisk/branches/10
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r369386 | mjordan | 2012-06-26 00:43:26 +0400 (Tue, 26 Jun 2012) | 5 lines
Update "manager show event" to support tab completion
Thank you rmudgett for pointing out that I was missing this in the initial
check-in for AMI event documentation (r369346)
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r369392 | mjordan | 2012-06-26 17:23:12 +0400 (Tue, 26 Jun 2012) | 18 lines
Fix crash in unloading of res_adsi module
When res_adsi is unloaded, it removes the ADSI functions that it previously installed
by passing a NULL adsi_funcs pointer to ast_adsi_install_funcs. This function was not
checking whether or not the adsi_funcs pointer passed in was NULL before dereferencing
it to check whether or not the version of the functions matches what the core was
expecting it.
This patch makes it so that the version is only checked if a potentially valid adsi_funcs
pointer was passed in. Passing in NULL removes the installed functions, bypassing the
version check.
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Merged revisions 369390 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 369391 from http://svn.asterisk.org/svn/asterisk/branches/10
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r369414 | jrose | 2012-06-27 01:45:22 +0400 (Wed, 27 Jun 2012) | 12 lines
Unique Call ID logging Phases III and IV
Adds call ID logging changes to specific channel drivers that weren't handled
handled in phase II of Call ID Logging. Also covers logging for threads for
threads created by systems that may be involved with many different calls.
Extra special thanks to Richard for rigorous review of chan_dahdi and its
various signalling modules.
review: https://reviewboard.asterisk.org/r/1927/
review: https://reviewboard.asterisk.org/r/1950/
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r369449 | twilson | 2012-06-28 01:21:27 +0400 (Thu, 28 Jun 2012) | 19 lines
Clean up after a reinvite that never gets a final response
The basic problem is that if a re-INVITE is sent by Asterisk and it receives a
provisional response, but no final response, then the dialog is never torn
down. In addition to leaking memory, this also leaks file descriptors and will
eventually lead to Asterisk no longer being able to process calls.
This patch just keeps track of whether there is an outstanding re-INVITE, and if
there is goes ahead and cleans up everything as though there was no outstanding
reinvite.
(closes issue ASTERISK-19992)
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Merged revisions 369436 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 369437 from http://svn.asterisk.org/svn/asterisk/branches/10
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r369454 | twilson | 2012-06-28 05:12:06 +0400 (Thu, 28 Jun 2012) | 22 lines
Add the ability to set flags via the config options api
Allows the setting of flags via the config options api.
For example, code like this:
#define OPT1 1 << 0
#define OPT2 1 << 1
#define OPT3 1 << 2
struct thing {
unsigned int flags;
};
and a config like this:
[blah]
opt1=yes
opt2=no
opt3=yes
Review: https://reviewboard.asterisk.org/r/2004/
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r369473 | file | 2012-06-29 19:33:39 +0400 (Fri, 29 Jun 2012) | 13 lines
Make the address family filter specific to the transport.
(closes issue ASTERISK-16618)
Reported by: Leif Madsen
Review: https://reviewboard.asterisk.org/r/1667/
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Merged revisions 369471 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 369472 from http://svn.asterisk.org/svn/asterisk/branches/10
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r369489 | rmudgett | 2012-06-29 20:42:32 +0400 (Fri, 29 Jun 2012) | 8 lines
Remove obsolete struct ast_channel note.
The opaquing the ast_channel struct no longer requires .cleancount to be
changed when the struct is changed.
* Bump .cleancount value one last time because of struct ast_channel for
old times sake.
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r369492 | file | 2012-06-29 20:56:29 +0400 (Fri, 29 Jun 2012) | 8 lines
With some configurations a transport is not actually specified so assume UDP in these cases.
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r369493 | rmudgett | 2012-06-29 21:02:32 +0400 (Fri, 29 Jun 2012) | 16 lines
Hangup handlers - Dialplan subroutines that run when the channel hangs up.
Hangup handlers are an alternative to the h extension. They can be used
in addition to the h extension. The idea is to attach a Gosub routine to
a channel that will execute when the call hangs up. Whereas which h
extension gets executed depends on the location of dialplan execution when
the call hangs up, hangup handlers are attached to the call channel. You
can attach multiple handlers that will execute in the order of most
recently added first.
(closes issue ASTERISK-19549)
Reported by: Mark Murawski
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/2002/
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Merged revisions 369013,369027-369028,369034,369057,369061,369068,369086,369092,369110,369126,369141-369142,369148,369166-369167,369184-369185,369190,369207,369216-369217,369237,369240,369260-369261,369264,369295-369296,369304,369326,369329,369346,369367,369370,369386,369392,369414,369449,369454,369473,369489,369492-369493 from http://svn.asterisk.org/svn/asterisk/trunk
Added:
team/may/ooh323_qsig/build_tools/find_missing_support_level
- copied unchanged from r369500, trunk/build_tools/find_missing_support_level
team/may/ooh323_qsig/build_tools/get_documentation.py
- copied unchanged from r369500, trunk/build_tools/get_documentation.py
team/may/ooh323_qsig/build_tools/post_process_documentation.py
- copied unchanged from r369500, trunk/build_tools/post_process_documentation.py
team/may/ooh323_qsig/main/adsi.c
- copied unchanged from r369500, trunk/main/adsi.c
Removed:
team/may/ooh323_qsig/res/res_adsi.exports.in
Modified:
team/may/ooh323_qsig/ (props changed)
team/may/ooh323_qsig/.cleancount
team/may/ooh323_qsig/CHANGES
team/may/ooh323_qsig/Makefile
team/may/ooh323_qsig/addons/chan_ooh323.c
team/may/ooh323_qsig/addons/ooh323c/src/ooCalls.c
team/may/ooh323_qsig/addons/ooh323c/src/ooq931.c
team/may/ooh323_qsig/apps/app_celgenuserevent.c
team/may/ooh323_qsig/apps/app_chanspy.c
team/may/ooh323_qsig/apps/app_confbridge.c
team/may/ooh323_qsig/apps/app_dial.c
team/may/ooh323_qsig/apps/app_externalivr.c
team/may/ooh323_qsig/apps/app_followme.c
team/may/ooh323_qsig/apps/app_meetme.c
team/may/ooh323_qsig/apps/app_queue.c
team/may/ooh323_qsig/apps/app_stack.c
team/may/ooh323_qsig/apps/app_system.c
team/may/ooh323_qsig/apps/app_userevent.c
team/may/ooh323_qsig/apps/app_voicemail.c
team/may/ooh323_qsig/apps/confbridge/conf_config_parser.c
team/may/ooh323_qsig/channels/chan_agent.c
team/may/ooh323_qsig/channels/chan_dahdi.c
team/may/ooh323_qsig/channels/chan_iax2.c
team/may/ooh323_qsig/channels/chan_local.c
team/may/ooh323_qsig/channels/chan_misdn.c
team/may/ooh323_qsig/channels/chan_sip.c
team/may/ooh323_qsig/channels/chan_skinny.c
team/may/ooh323_qsig/channels/console_board.c
team/may/ooh323_qsig/channels/console_gui.c
team/may/ooh323_qsig/channels/console_video.c
team/may/ooh323_qsig/channels/iax2-parser.c
team/may/ooh323_qsig/channels/iax2-provision.c
team/may/ooh323_qsig/channels/misdn/ie.c
team/may/ooh323_qsig/channels/misdn/isdn_lib.c
team/may/ooh323_qsig/channels/misdn/isdn_msg_parser.c
team/may/ooh323_qsig/channels/misdn/portinfo.c
team/may/ooh323_qsig/channels/misdn_config.c
team/may/ooh323_qsig/channels/sig_analog.c
team/may/ooh323_qsig/channels/sig_pri.c
team/may/ooh323_qsig/channels/sig_ss7.c
team/may/ooh323_qsig/channels/sip/config_parser.c
team/may/ooh323_qsig/channels/sip/dialplan_functions.c
team/may/ooh323_qsig/channels/sip/include/sip.h
team/may/ooh323_qsig/channels/sip/reqresp_parser.c
team/may/ooh323_qsig/channels/sip/sdp_crypto.c
team/may/ooh323_qsig/channels/sip/security_events.c
team/may/ooh323_qsig/channels/sip/srtp.c
team/may/ooh323_qsig/channels/sip/utils.c
team/may/ooh323_qsig/channels/vcodecs.c
team/may/ooh323_qsig/channels/vgrabbers.c
team/may/ooh323_qsig/codecs/codec_dahdi.c
team/may/ooh323_qsig/configs/cdr.conf.sample
team/may/ooh323_qsig/configs/config_test.conf.sample
team/may/ooh323_qsig/configure
team/may/ooh323_qsig/configure.ac
team/may/ooh323_qsig/doc/appdocsxml.dtd
team/may/ooh323_qsig/funcs/func_channel.c
team/may/ooh323_qsig/funcs/func_presencestate.c
team/may/ooh323_qsig/funcs/func_realtime.c
team/may/ooh323_qsig/include/asterisk/adsi.h
team/may/ooh323_qsig/include/asterisk/channel.h
team/may/ooh323_qsig/include/asterisk/config_options.h
team/may/ooh323_qsig/include/asterisk/logger.h
team/may/ooh323_qsig/include/asterisk/netsock2.h
team/may/ooh323_qsig/include/asterisk/pbx.h
team/may/ooh323_qsig/include/asterisk/xmldoc.h
team/may/ooh323_qsig/main/Makefile
team/may/ooh323_qsig/main/abstract_jb.c
team/may/ooh323_qsig/main/acl.c
team/may/ooh323_qsig/main/alaw.c
team/may/ooh323_qsig/main/aoc.c
team/may/ooh323_qsig/main/app.c
team/may/ooh323_qsig/main/asterisk.c
team/may/ooh323_qsig/main/astfd.c
team/may/ooh323_qsig/main/astmm.c
team/may/ooh323_qsig/main/astobj2.c
team/may/ooh323_qsig/main/audiohook.c
team/may/ooh323_qsig/main/autochan.c
team/may/ooh323_qsig/main/autoservice.c
team/may/ooh323_qsig/main/bridging.c
team/may/ooh323_qsig/main/callerid.c
team/may/ooh323_qsig/main/ccss.c
team/may/ooh323_qsig/main/cdr.c
team/may/ooh323_qsig/main/cel.c
team/may/ooh323_qsig/main/channel.c
team/may/ooh323_qsig/main/channel_internal_api.c
team/may/ooh323_qsig/main/chanvars.c
team/may/ooh323_qsig/main/cli.c
team/may/ooh323_qsig/main/config.c
team/may/ooh323_qsig/main/config_options.c
team/may/ooh323_qsig/main/data.c
team/may/ooh323_qsig/main/datastore.c
team/may/ooh323_qsig/main/db.c
team/may/ooh323_qsig/main/devicestate.c
team/may/ooh323_qsig/main/dial.c
team/may/ooh323_qsig/main/dns.c
team/may/ooh323_qsig/main/dnsmgr.c
team/may/ooh323_qsig/main/dsp.c
team/may/ooh323_qsig/main/enum.c
team/may/ooh323_qsig/main/event.c
team/may/ooh323_qsig/main/features.c
team/may/ooh323_qsig/main/file.c
team/may/ooh323_qsig/main/fixedjitterbuf.c
team/may/ooh323_qsig/main/format.c
team/may/ooh323_qsig/main/format_cap.c
team/may/ooh323_qsig/main/format_pref.c
team/may/ooh323_qsig/main/frame.c
team/may/ooh323_qsig/main/framehook.c
team/may/ooh323_qsig/main/fskmodem.c
team/may/ooh323_qsig/main/fskmodem_float.c
team/may/ooh323_qsig/main/fskmodem_int.c
team/may/ooh323_qsig/main/global_datastores.c
team/may/ooh323_qsig/main/hashtab.c
team/may/ooh323_qsig/main/heap.c
team/may/ooh323_qsig/main/image.c
team/may/ooh323_qsig/main/indications.c
team/may/ooh323_qsig/main/io.c
team/may/ooh323_qsig/main/jitterbuf.c
team/may/ooh323_qsig/main/libasteriskssl.c
team/may/ooh323_qsig/main/loader.c
team/may/ooh323_qsig/main/lock.c
team/may/ooh323_qsig/main/logger.c
team/may/ooh323_qsig/main/manager.c
team/may/ooh323_qsig/main/message.c
team/may/ooh323_qsig/main/netsock.c
team/may/ooh323_qsig/main/netsock2.c
team/may/ooh323_qsig/main/pbx.c
team/may/ooh323_qsig/main/plc.c
team/may/ooh323_qsig/main/presencestate.c
team/may/ooh323_qsig/main/privacy.c
team/may/ooh323_qsig/main/rtp_engine.c
team/may/ooh323_qsig/main/say.c
team/may/ooh323_qsig/main/sched.c
team/may/ooh323_qsig/main/security_events.c
team/may/ooh323_qsig/main/slinfactory.c
team/may/ooh323_qsig/main/srv.c
team/may/ooh323_qsig/main/stdtime/localtime.c
team/may/ooh323_qsig/main/strcompat.c
team/may/ooh323_qsig/main/strings.c
team/may/ooh323_qsig/main/stun.c
team/may/ooh323_qsig/main/syslog.c
team/may/ooh323_qsig/main/taskprocessor.c
team/may/ooh323_qsig/main/tcptls.c
team/may/ooh323_qsig/main/tdd.c
team/may/ooh323_qsig/main/term.c
team/may/ooh323_qsig/main/test.c
team/may/ooh323_qsig/main/threadstorage.c
team/may/ooh323_qsig/main/timing.c
team/may/ooh323_qsig/main/translate.c
team/may/ooh323_qsig/main/udptl.c
team/may/ooh323_qsig/main/ulaw.c
team/may/ooh323_qsig/main/utils.c
team/may/ooh323_qsig/main/xml.c
team/may/ooh323_qsig/main/xmldoc.c
team/may/ooh323_qsig/makeopts.in
team/may/ooh323_qsig/pbx/dundi-parser.c
team/may/ooh323_qsig/res/ael/pval.c
team/may/ooh323_qsig/res/res_adsi.c
team/may/ooh323_qsig/res/snmp/agent.c
team/may/ooh323_qsig/tests/test_config.c
team/may/ooh323_qsig/utils/astdb2bdb.c
team/may/ooh323_qsig/utils/astdb2sqlite3.c
Propchange: team/may/ooh323_qsig/
------------------------------------------------------------------------------
--- branch-10-digiumphones-merged (original)
+++ branch-10-digiumphones-merged Tue Jul 3 05:52:25 2012
@@ -1,1 +1,1 @@
-/branches/10-digiumphones:364766,365396,368791,368963-368965
+/branches/10-digiumphones:364766,365396,368791,368963-368965,368999,369026
Propchange: team/may/ooh323_qsig/
------------------------------------------------------------------------------
Binary property 'branch-10-merged' - no diff available.
Propchange: team/may/ooh323_qsig/
------------------------------------------------------------------------------
--- svn:mergeinfo (original)
+++ svn:mergeinfo Tue Jul 3 05:52:25 2012
@@ -1,2 +1,2 @@
/team/mmichelson/private/phones-trunk:358764-361321
-/trunk:357542,368181-369007
+/trunk:357542,368181-369500
Propchange: team/may/ooh323_qsig/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Tue Jul 3 05:52:25 2012
@@ -1,1 +1,1 @@
-/trunk:1-368158,368181-369007
+/trunk:1-368158,368181-369500
Modified: team/may/ooh323_qsig/.cleancount
URL: http://svnview.digium.com/svn/asterisk/team/may/ooh323_qsig/.cleancount?view=diff&rev=369549&r1=369548&r2=369549
==============================================================================
--- team/may/ooh323_qsig/.cleancount (original)
+++ team/may/ooh323_qsig/.cleancount Tue Jul 3 05:52:25 2012
@@ -1,3 +1,1 @@
-39
-
-
+40
Modified: team/may/ooh323_qsig/CHANGES
URL: http://svnview.digium.com/svn/asterisk/team/may/ooh323_qsig/CHANGES?view=diff&rev=369549&r1=369548&r2=369549
==============================================================================
--- team/may/ooh323_qsig/CHANGES (original)
+++ team/may/ooh323_qsig/CHANGES Tue Jul 3 05:52:25 2012
@@ -11,6 +11,14 @@
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
------------------------------------------------------------------------------
+
+Build System
+----
+ * A new make target, 'full', has been added to the Makefile. This performs
+ the same compilation actions as make all, but will also scan the entirety of
+ each source file for documentation. This option is needed to generate AMI
+ event documentation. Note that your system must have Python in order for
+ this make target to succeed.
Core
----
@@ -38,6 +46,9 @@
have their own verbosity level. The command 'core set verbose' will now set
a separate level for each remote console without affecting any other
console.
+ * Hangup handlers can be attached to channels using the CHANNEL(hangup_handler_xxx)
+ options. Hangup handlers will run when the channel is hung up similar to the
+ h extension.
CLI Changes
-------------------
@@ -199,6 +210,15 @@
to hangup multiple channels, use /regex/ as the Channel option. Existing
behavior to hanging up a single channel is unchanged, but if you pass a regex,
the manager will send you a list of channels back that were hung up.
+
+ * Support for IPv6 addresses has been added.
+
+ * AMI Events can now be documented in the Asterisk source. Two new CLI
+ commands have been added to display information about AMI events at run time:
+ manager show events, which shows a list of all known and documented AMI
+ events, and manager show event [event name], which shows detail information
+ about a specific AMI event. Note that AMI event documentation is only
+ generated when Asterisk is compiled using 'make full'.
FAX changes
-----------
Modified: team/may/ooh323_qsig/Makefile
URL: http://svnview.digium.com/svn/asterisk/team/may/ooh323_qsig/Makefile?view=diff&rev=369549&r1=369548&r2=369549
==============================================================================
--- team/may/ooh323_qsig/Makefile (original)
+++ team/may/ooh323_qsig/Makefile Tue Jul 3 05:52:25 2012
@@ -96,6 +96,7 @@
export WGET_EXTRA_ARGS
export LDCONFIG
export LDCONFIG_FLAGS
+export PYTHON
# even though we could use '-include makeopts' here, use a wildcard
# lookup anyway, so that make won't try to build makeopts if it doesn't
@@ -315,12 +316,28 @@
@echo " + $(mK) install +"
@echo " +-------------------------------------------+"
+full: _cleantest_all_full
+ @echo " +--------- Asterisk Build Complete ---------+"
+ @echo " + Asterisk has successfully been built, and +"
+ @echo " + can be installed by running: +"
+ @echo " + +"
+ @echo " + $(mK) install +"
+ @echo " +-------------------------------------------+"
+
+
# For parallel builds, we must call cleantest *before* running the
# other dependencies on _all.
_cleantest_all: cleantest
@$(MAKE) _all
+# For parallel builds, we must call cleantest *before* running the
+# other dependencies on _all.
+_cleantest_all_full: cleantest
+ @$(MAKE) _all_full
+
_all: makeopts $(SUBDIRS) doc/core-en_US.xml $(ADDL_TARGETS)
+
+_all_full: makeopts $(SUBDIRS) doc/full-en_US.xml $(ADDL_TARGETS)
makeopts: configure
@echo "****"
@@ -413,6 +430,8 @@
rm -f defaults.h
rm -f include/asterisk/build.h
rm -f main/version.c
+ rm -f doc/core-en_US.xml
+ rm -f doc/full-en_US.xml
@$(MAKE) -C menuselect clean
cp -f .cleancount .lastclean
@@ -464,6 +483,27 @@
done
@echo
@echo "</docs>" >> $@
+
+doc/full-en_US.xml: $(foreach dir,$(MOD_SUBDIRS),$(shell $(GREP) -l "language=\"en_US\"" $(dir)/*.c $(dir)/*.cc 2>/dev/null))
+ifeq ($(PYTHON),:)
+ @echo "--------------------------------------------------------------------------"
+ @echo "--- Please install python to build full documentation ---"
+ @echo "--------------------------------------------------------------------------"
+else
+ @printf "Building Documentation For: "
+ @echo "<?xml version=\"1.0\" encoding=\"UTF-8\"?>" > $@
+ @echo "<!DOCTYPE docs SYSTEM \"appdocsxml.dtd\">" >> $@
+ @echo "<docs xmlns:xi=\"http://www.w3.org/2001/XInclude\">" >> $@
+ @for x in $(MOD_SUBDIRS); do \
+ printf "$$x " ; \
+ for i in $$x/*.c; do \
+ $(PYTHON) build_tools/get_documentation.py < $$i >> $@ ; \
+ done ; \
+ done
+ @echo
+ @echo "</docs>" >> $@
+ @$(PYTHON) build_tools/post_process_documentation.py -i $@ -o "doc/core-en_US.xml"
+endif
validate-docs: doc/core-en_US.xml
ifeq ($(XMLSTARLET)$(XMLLINT),::)
Modified: team/may/ooh323_qsig/addons/chan_ooh323.c
URL: http://svnview.digium.com/svn/asterisk/team/may/ooh323_qsig/addons/chan_ooh323.c?view=diff&rev=369549&r1=369548&r2=369549
==============================================================================
--- team/may/ooh323_qsig/addons/chan_ooh323.c (original)
+++ team/may/ooh323_qsig/addons/chan_ooh323.c Tue Jul 3 05:52:25 2012
@@ -1158,6 +1158,8 @@
if (p) {
ast_mutex_lock(&p->lock);
+
+ p->lastrtptx = time(NULL);
[... 10919 lines stripped ...]
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