[asterisk-commits] eliel: branch group/data_api_gsoc2009 r219804 - in /team/group/data_api_gsoc2...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Sep 22 09:43:51 CDT 2009
Author: eliel
Date: Tue Sep 22 09:43:42 2009
New Revision: 219804
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=219804
Log:
Merged revisions 216955-216956,216993,217015,217033,217074,217113,217158,217199,217236,217286,217331-217332,217367-217368,217408,217445,217482,217524,217560,217593,217638,217663,217669,217730,217737,217744,217804,217807,217873,217912,217916,217918,217954,217987,217990,218050,218107,218150,218184,218224,218295,218361,218365,218430,218465,218499-218500,218504,218566,218579,218583,218586,218629,218687,218731,218799,218868,218918,218933,218973,219007,219061,219105,219139,219230,219264,219304,219324,219371,219412,219451,219520,219587,219654,219721 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r216955 | oej | 2009-09-07 16:19:37 -0400 (Mon, 07 Sep 2009) | 2 lines
Add new actions under "new actions" and not in the top of the document
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r216956 | oej | 2009-09-07 16:23:19 -0400 (Mon, 07 Sep 2009) | 2 lines
Fixing formatting
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r216993 | dvossel | 2009-09-08 10:26:30 -0400 (Tue, 08 Sep 2009) | 14 lines
caller id number empty
parse_uri was not being given the correct scheme's, as
a result, uri parsing did not parse the username correctly.
One of the side effects of this is an empty caller id.
(closes issue #15839)
Reported by: ebroad
Patches:
blank_cidv2.patch uploaded by ebroad (license 878)
parse_uri_fix.diff uploaded by dvossel (license 671)
Tested by: ebroad, dvossel
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r217015 | tzafrir | 2009-09-08 11:23:04 -0400 (Tue, 08 Sep 2009) | 8 lines
live_ast: Fix asterisk.conf instead of regenerating it
* Don't write asterisk.conf from scratch. Fix the existing one.
* Pass extra 'make' command-line arguments to 'install' and 'samples'.
* Fix some extra typos.
closes issue #15019 .
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r217033 | tilghman | 2009-09-08 11:30:18 -0400 (Tue, 08 Sep 2009) | 4 lines
Remove what appears to be an unnecessary define.
(closes issue #15851)
Reported by: tzafrir
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r217074 | kpfleming | 2009-09-08 12:37:28 -0400 (Tue, 08 Sep 2009) | 9 lines
Ensure that the default autoconf CFLAGS are not used.
A recent change to the configure script that allows the user to specify
CFLAGS and/or LDFLAGS to the script had the unfortunate side effect of
letting autoconf's default CFLAGS (-g -O2) feed in to the rest of the build
system, thereby overriding the DONT_OPTIMIZE setting in menuselect. That
problem is now corrected.
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r217113 | russell | 2009-09-08 14:06:57 -0400 (Tue, 08 Sep 2009) | 13 lines
Fix audio problems with format_mp3.
This problem was introduced when the AST_FRIENDLY_OFFSET patch was merged.
I'm surprised that nobody noticed any trouble when testing that patch, but this
fixes the code that fills in the buffer to start filling in after the offset
portion of the buffer.
(closes issue #15850)
Reported by: 99gixxer
Patches:
issue15850.diff1.txt uploaded by russell (license 2)
Tested by: 99gixxer
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r217158 | mmichelson | 2009-09-08 16:06:15 -0400 (Tue, 08 Sep 2009) | 6 lines
Add doxygen to ast_event_subscribe for the description.
Most importantly, note that a NULL description will cause a
crash, as I just experienced that firsthand.
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r217199 | tilghman | 2009-09-08 16:28:41 -0400 (Tue, 08 Sep 2009) | 14 lines
Merged revisions 217156 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r217156 | tilghman | 2009-09-08 15:01:45 -0500 (Tue, 08 Sep 2009) | 7 lines
When MOH is playing on the channel, announcements sent through the conference are not heard.
(closes issue #14588)
Reported by: voipas
Patches:
20090716__issue14588__2.diff.txt uploaded by tilghman (license 14)
Tested by: lmadsen, twisted, tilghman
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r217236 | rmudgett | 2009-09-08 17:17:16 -0400 (Tue, 08 Sep 2009) | 1 line
Remove duplicate entry in the sig_pri_pri private pointer array.
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r217286 | seanbright | 2009-09-08 18:17:08 -0400 (Tue, 08 Sep 2009) | 4 lines
Fix compilation of app_meetme.
Reported by ebroad in #asterisk-bugs
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r217331 | rmudgett | 2009-09-08 19:31:27 -0400 (Tue, 08 Sep 2009) | 1 line
Miscellaneous minor code cleanup in mkintf().
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r217332 | rmudgett | 2009-09-08 19:37:57 -0400 (Tue, 08 Sep 2009) | 1 line
Fix memory leak of sig_xxx private structures.
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r217367 | oej | 2009-09-09 06:38:45 -0400 (Wed, 09 Sep 2009) | 2 lines
Formatting and doxygen updates
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r217368 | oej | 2009-09-09 06:39:43 -0400 (Wed, 09 Sep 2009) | 2 lines
Not having any TLS session to write to is a serious XMIT_ERROR.
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r217408 | seanbright | 2009-09-09 08:11:12 -0400 (Wed, 09 Sep 2009) | 8 lines
Properly terminate the response to the manager Ping action.
In passing, correct the formatting of the Timestamp attribute so that there is a
space after the colon and before the value.
(closes issue #15861)
Reported by: Ivan
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r217445 | tzafrir | 2009-09-09 14:52:48 -0400 (Wed, 09 Sep 2009) | 6 lines
gcc 4.4 fix: union instead of cast
gcc 4.4 has more strict rules for aliasing. It doesn't like a
struct sockaddr_in pointer pointing to a struct sockaddr. So we make it
a union.
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r217482 | oej | 2009-09-09 16:09:31 -0400 (Wed, 09 Sep 2009) | 9 lines
Don't report transfer success until we actually know. 1xx messages are not final.
Related to #12713
Patch by oej
A big thank you to file for finally fixing the transfer() dialplan application.
I've been waiting for years for this. Great work!
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r217524 | moy | 2009-09-09 17:48:04 -0400 (Wed, 09 Sep 2009) | 1 line
ast_log replaced for ast_verbose in MFCR2 event notifications
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r217560 | rmudgett | 2009-09-09 20:35:30 -0400 (Wed, 09 Sep 2009) | 1 line
Fix available() for SS7, MFC/R2, and pseudo channels.
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r217593 | oej | 2009-09-10 08:06:55 -0400 (Thu, 10 Sep 2009) | 8 lines
Include ActionID in all events that are responsed to AMI Action SIPShowRegistry
(closes issue #15868)
Reported by: nic_bellamy
Patches:
manager_SIPshowregistry_actionid.patch uploaded by nic bellamy (license 299)
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r217638 | tilghman | 2009-09-10 14:17:14 -0400 (Thu, 10 Sep 2009) | 4 lines
Verify support for wide ODBC character types before using them.
(closes issue #15870)
Reported by: nic_bellamy
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r217663 | oej | 2009-09-10 14:29:21 -0400 (Thu, 10 Sep 2009) | 2 lines
Don't assign UINT_MAX to an INT.
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r217669 | oej | 2009-09-10 15:09:02 -0400 (Thu, 10 Sep 2009) | 16 lines
Blocked revisions 217668 via svnmerge
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r217668 | oej | 2009-09-10 21:07:24 +0200 (Tor, 10 Sep 2009) | 9 lines
Remove harmful code that causes endless loops.
Remove code that causes loops in registrations.
We have agreed that the patch that this code was part of was bad. I am ripping out the code that causes
the issue. putnopvut needs to check the rest of the patch, if it needs to be changed as well.
This solves the issue reported in #15540, but needs more work before we close it (as described above).
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r217730 | mnick | 2009-09-10 15:39:41 -0400 (Thu, 10 Sep 2009) | 17 lines
Sets the correct musicclass after an announcement
(closes issue #15279)
Reported by: mbeckwell
Patches:
patch.txt uploaded by mnick (license )
Tested by: mnick
(closes issue #15832)
Reported by: mbeckwell
Patches:
patch.txt uploaded by mnick (license 874)
Tested by: mnick
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r217737 | oej | 2009-09-10 15:55:16 -0400 (Thu, 10 Sep 2009) | 11 lines
Blocked revisions 217735 via svnmerge
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r217735 | oej | 2009-09-10 21:52:19 +0200 (Tor, 10 Sep 2009) | 4 lines
Reinstate muted that was removed by mistake.
muted doesn't compile any more on os/x, so I have to disable it in order to testcompile other code...
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r217744 | jpeeler | 2009-09-10 16:18:30 -0400 (Thu, 10 Sep 2009) | 7 lines
Stop caller id transmission when offhook event detected.
This fixes the problem that would occur if an analog phone was picked up while
the caller id was being sent. The caller id before sent the whole spill even
after pickup and is now corrected.
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r217804 | jpeeler | 2009-09-10 16:52:57 -0400 (Thu, 10 Sep 2009) | 5 lines
Fix crash during attended transfer over PRI.
The owner pointers in the sig_pri_chan structure were not getting updated
in dahdi_fixup.
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r217807 | dvossel | 2009-09-10 17:07:47 -0400 (Thu, 10 Sep 2009) | 28 lines
Merged revisions 217806 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r217806 | dvossel | 2009-09-10 16:06:07 -0500 (Thu, 10 Sep 2009) | 22 lines
IAX2 encryption regression
The IAX2 Call Token security patch inadvertently broke the use of
encryption due to the reorganization of code in the socket_process()
function. When encryption is used, an incoming full frame must first
be decrypted before the information elements can be parsed. The
security release mistakenly moved IE parsing before decryption in
order to process the new Call Token IE. To resolve this, decryption
of full frames is once again done before looking into the frame. This
involves searching for an existing callno, checking the pvt to see if
encryption is turned on, and decrypting the packet before the internal
fields of the full frame are accessed.
(closes issue #15834)
Reported by: karesmakro
Patches:
iax2_encryption_fix_1.4.diff uploaded by dvossel (license 671)
Tested by: dvossel, karesmakro
Review: https://reviewboard.asterisk.org/r/355/
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r217873 | rmudgett | 2009-09-10 18:11:17 -0400 (Thu, 10 Sep 2009) | 1 line
Miscellaneous minor changes.
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r217912 | rmudgett | 2009-09-10 18:31:12 -0400 (Thu, 10 Sep 2009) | 8 lines
Cleaned up chan_dahdi iflist handling and locking.
* Fixed walking the iflist so it is always done with the iflock locked.
* Simplified iflist walking routines.
* Created chan_dahdi iflist insertion and extraction routines.
* Fixed duplicate_pseudo() malloc fail handling.
* Fixed infinite loop in action_dahdishowchannels() when showing a single channel.
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r217916 | tilghman | 2009-09-10 19:12:16 -0400 (Thu, 10 Sep 2009) | 5 lines
Make calltoken support work with realtime users and peers.
In the course of this, I also found that the results of ast_gethostbyname
were being used incorrectly in both chan_iax2 and chan_sip, so both have
been fixed.
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r217918 | tilghman | 2009-09-10 19:16:24 -0400 (Thu, 10 Sep 2009) | 8 lines
Blocked revisions 217917 via svnmerge
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r217917 | tilghman | 2009-09-10 18:15:21 -0500 (Thu, 10 Sep 2009) | 2 lines
Backport realtime fix to 1.4
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r217954 | jpeeler | 2009-09-10 19:29:14 -0400 (Thu, 10 Sep 2009) | 2 lines
Allow do not disturb to be set on analog channels via the CLI and AMI.
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r217987 | jpeeler | 2009-09-10 19:49:09 -0400 (Thu, 10 Sep 2009) | 2 lines
Cleanup approach in 217804 and don't reach inside the sig_pvt.
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r217990 | tilghman | 2009-09-10 19:54:51 -0400 (Thu, 10 Sep 2009) | 10 lines
Merged revisions 217989 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r217989 | tilghman | 2009-09-10 18:52:22 -0500 (Thu, 10 Sep 2009) | 3 lines
Don't ring another channel, if there's not enough time for a queue member to answer.
(Fixes AST-228)
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r218050 | tilghman | 2009-09-11 01:58:11 -0400 (Fri, 11 Sep 2009) | 3 lines
Check the origination priority for more matches, not the current priority.
Found by Pavel Troller on the -dev list.
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r218107 | mvanbaak | 2009-09-12 09:08:16 -0400 (Sat, 12 Sep 2009) | 8 lines
use the actual given ip address for 'rtp set debug ip <foo>' instead of the word 'ip'
(closes issue #15711)
Reported by: davidw
Patches:
2009082800-rtpdebug.diff.txt uploaded by mvanbaak (license 7)
Tested by: davidw
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r218150 | moy | 2009-09-13 01:51:46 -0400 (Sun, 13 Sep 2009) | 1 line
get rid of mfcr2 monitor thread condition, is problematic
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r218184 | tzafrir | 2009-09-13 13:34:11 -0400 (Sun, 13 Sep 2009) | 5 lines
gcc 4.4: Remove a nop memset size 0 that annoys gcc
This memset doesn't write beyond the end of the buffer.
(tmpbuf has size of 4).
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r218224 | mnicholson | 2009-09-14 10:57:23 -0400 (Mon, 14 Sep 2009) | 14 lines
Merged revisions 218223 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r218223 | mnicholson | 2009-09-14 09:53:57 -0500 (Mon, 14 Sep 2009) | 8 lines
Ensure we don't pickup ourselves when doing pickup by exten.
(closes issue #15100)
Reported by: lmsteffan
Patches:
(modified) pickup.patch uploaded by lmsteffan (license 779)
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r218295 | file | 2009-09-14 14:16:39 -0400 (Mon, 14 Sep 2009) | 2 lines
Do not attempt to add a parking extension if an error occurred while reading the configuration.
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r218361 | tilghman | 2009-09-14 15:29:48 -0400 (Mon, 14 Sep 2009) | 11 lines
Recorded merge of revisions 218331 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r218331 | tilghman | 2009-09-14 14:16:35 -0500 (Mon, 14 Sep 2009) | 4 lines
Don't say "Please try again" if we don't give the user another chance to try again.
(issue #15055, SWP-129)
Reported by: jthurman
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r218365 | rmudgett | 2009-09-14 16:08:11 -0400 (Mon, 14 Sep 2009) | 4 lines
Add support for multiple interface lists.
Also unlink the sig_pri_pri.pvts[] pointer in destroy_dahdi_pvt().
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r218430 | jpeeler | 2009-09-14 18:38:25 -0400 (Mon, 14 Sep 2009) | 18 lines
Merged revisions 218401 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r218401 | jpeeler | 2009-09-14 16:47:11 -0500 (Mon, 14 Sep 2009) | 11 lines
Fix handling of DAHDI_EVENT_REMOVED event to prevent crash in do_monitor.
After talking to rmudgett about some of his recent iflist locking changes, it
was determined that the only place that would destroy a channel without being
explicitly to do so was in handle_init_event. The loop to walk the interface
list has been modified to wait to destroy the channel until the dahdi_pvt of
the channel to be destroyed is no longer needed.
(closes issue #15378)
Reported by: samy
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r218465 | tzafrir | 2009-09-15 06:24:55 -0400 (Tue, 15 Sep 2009) | 2 lines
Fix false error message on DAHDI_EVENT_REMOVED (RESULT_SUCCESS == 0)
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r218499 | mmichelson | 2009-09-15 10:59:50 -0400 (Tue, 15 Sep 2009) | 3 lines
Fix off-by-one error when reading SDP sent over TCP.
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r218500 | kpfleming | 2009-09-15 11:02:21 -0400 (Tue, 15 Sep 2009) | 9 lines
Merged revisions 218497 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r218497 | kpfleming | 2009-09-15 10:55:58 -0400 (Tue, 15 Sep 2009) | 1 line
Use proper hostname for downloading sound files.
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r218504 | mmichelson | 2009-09-15 11:05:53 -0400 (Tue, 15 Sep 2009) | 3 lines
Ensure that SDP read from TCP socket is null-terminated.
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r218566 | mmichelson | 2009-09-15 11:40:14 -0400 (Tue, 15 Sep 2009) | 4 lines
Use a better method of ensuring null-termination of the buffer
while reading the SDP when using TCP.
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r218579 | tilghman | 2009-09-15 12:04:41 -0400 (Tue, 15 Sep 2009) | 16 lines
Merged revisions 218577 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r218577 | tilghman | 2009-09-15 11:01:17 -0500 (Tue, 15 Sep 2009) | 9 lines
Ensure FollowMe sets language in channels it creates.
Also, not in the original bug report, but related fields are accountcode and
musicclass, and the inheritance of datastores.
(closes issue #15372)
Reported by: Romik
Patches:
20090828__issue15372.diff.txt uploaded by tilghman (license 14)
Tested by: cervajs
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r218583 | jpeeler | 2009-09-15 12:12:49 -0400 (Tue, 15 Sep 2009) | 5 lines
Add some changes related to 218430.
* Remove thread_spawned in handle_init_event since it was never used
* Always check handle_init_event in case a channel is destroyed
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r218586 | mnicholson | 2009-09-15 12:15:02 -0400 (Tue, 15 Sep 2009) | 15 lines
Merged revisions 218578 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r218578 | mnicholson | 2009-09-15 11:03:54 -0500 (Tue, 15 Sep 2009) | 8 lines
Send request contact header field with response to registrer queries instead of the address of record.
(closes issue #14438)
Reported by: ravindrad
Patches:
regquerypatch uploaded by ravindrad (license 684)
Tested by: ravindrad
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r218629 | jpeeler | 2009-09-15 12:30:43 -0400 (Tue, 15 Sep 2009) | 9 lines
Blocked revisions 218623 via svnmerge
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r218623 | jpeeler | 2009-09-15 11:29:27 -0500 (Tue, 15 Sep 2009) | 3 lines
Fix small memory leak in handle_init_event by always destroying the pthread
attr before returning.
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r218687 | dvossel | 2009-09-15 15:22:37 -0400 (Tue, 15 Sep 2009) | 2 lines
upward bound checking for port string to int conversion
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r218731 | tilghman | 2009-09-15 18:33:10 -0400 (Tue, 15 Sep 2009) | 13 lines
Merged revisions 218730 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r218730 | tilghman | 2009-09-15 17:27:41 -0500 (Tue, 15 Sep 2009) | 6 lines
If the user enters the same password as before, don't signal an error when the change does nothing.
(closes issue #15492)
Reported by: cbbs70a
Patches:
20090713__issue15492.diff.txt uploaded by tilghman (license 14)
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r218799 | russell | 2009-09-16 09:34:41 -0400 (Wed, 16 Sep 2009) | 16 lines
Merged revisions 218798 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r218798 | russell | 2009-09-16 08:33:43 -0500 (Wed, 16 Sep 2009) | 9 lines
Remove the IAXy firmware from Asterisk.
The firmware can now be found on downloads.digium.com, where the rest of our
binary downloads live. This was the last part of our Asterisk tarballs that
was considered non-free by Debian. :-)
(closes issue #15838)
Reported by: paravoid
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r218868 | dbrooks | 2009-09-16 14:06:42 -0400 (Wed, 16 Sep 2009) | 20 lines
Merged revisions 218867 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r218867 | dbrooks | 2009-09-16 13:00:45 -0500 (Wed, 16 Sep 2009) | 13 lines
Fixes CID pattern matching behavior to mirror that of extension pattern matching.
Pattern matching for extensions uses a type of scoring system, giving values for
specificity to each character in the pattern. Unfortunately, this is done character
by character, in order. This does lead to some less specific patterns being first
in line for matching, but it will usually get the job done.
This patch merely brings CID matching to the same level as extension matching.
This patch does not attempt to tackle the problem shared by extension matching.
(closes issue #14708)
Reported by: klaus3000
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r218918 | file | 2009-09-16 14:31:47 -0400 (Wed, 16 Sep 2009) | 5 lines
On TCP and TLS connections do not attempt to stop retransmission of the packet internally.
This was preventing responses from being properly processed because the packet was not being found
causing handle_response to return prematurely.
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r218933 | mmichelson | 2009-09-16 15:25:36 -0400 (Wed, 16 Sep 2009) | 12 lines
Reverse order of args to fread.
This way, we don't always write a null byte into
byte 1 of the buffer
(closes issue #15905)
Reported by: ebroad
Patches:
freadfix.patch uploaded by ebroad (license 878)
Tested by: ebroad
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r218973 | seanbright | 2009-09-16 16:32:50 -0400 (Wed, 16 Sep 2009) | 7 lines
Remove some unused defines from res_jabber.
(closes issue #15359)
Reported by: snuffy
Patches:
bug_res_jabber_unused_defines.diff uploaded by snuffy (license 35)
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r219007 | tilghman | 2009-09-16 19:15:43 -0400 (Wed, 16 Sep 2009) | 7 lines
Detect whether we actually have the long double type, before looking for those functions.
(closes issue #15017)
Reported by: tzafrir
Patches:
20090916__issue15017.diff.txt uploaded by tilghman (license 14)
Tested by: tzafrir
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r219061 | tilghman | 2009-09-16 19:42:12 -0400 (Wed, 16 Sep 2009) | 15 lines
Merged revisions 219023 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r219023 | tilghman | 2009-09-16 18:21:53 -0500 (Wed, 16 Sep 2009) | 8 lines
Properly deal with quotes in the arguments of '#exec' includes.
(closes issue #15583)
Reported by: pkempgen
Patches:
20090726__issue15583.diff.txt uploaded by tilghman (license 14)
20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license 169)
Tested by: pkempgen
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r219105 | tilghman | 2009-09-16 20:58:10 -0400 (Wed, 16 Sep 2009) | 10 lines
Add the 'E' option to exit ChanSpy, once the single channel it spied upon hangs up.
In addition, there's a bit of cleanup to the arguments and documentation, in which
I discovered that the last feature added to this application duplicated an option
(oops!) and changed that option so that it now works.
(closes issue #14909)
Reported by: junky
Patches:
__20090901-spy_hangup_trunk.diff uploaded by lmadsen (license 10)
Tested by: amilcar, junky, flujan, lmadsen
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r219139 | mnicholson | 2009-09-17 11:18:01 -0400 (Thu, 17 Sep 2009) | 17 lines
Merged revisions 219136 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r219136 | mnicholson | 2009-09-17 09:58:39 -0500 (Thu, 17 Sep 2009) | 10 lines
Prevent a potential race condition and crash when hanging up a channel by removing the channel from the channel list before begining channel tear down.
This fix may potentially cause problems with CDR backends that access the channel a CDR is associated with via the channel list. This fix makes the channel unavabile at the time when the CDR backend is invoked. This has been documented in include/asterisk/cdr.h.
(closes issue #15316)
Reported by: vmarrone
Tested by: mnicholson
Review: https://reviewboard.asterisk.org/r/362/
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r219230 | seanbright | 2009-09-17 12:25:38 -0400 (Thu, 17 Sep 2009) | 2 lines
Get this compiling under dev-mode.
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r219264 | file | 2009-09-17 15:57:39 -0400 (Thu, 17 Sep 2009) | 2 lines
Ensure no spaces exist before "refresher=" when doing the comparison.
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r219304 | dvossel | 2009-09-17 17:59:21 -0400 (Thu, 17 Sep 2009) | 27 lines
Merged revisions 219303 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r219303 | dvossel | 2009-09-17 16:29:37 -0500 (Thu, 17 Sep 2009) | 21 lines
INVITE w/Replaces deadlock fix
This patch cleans up the locking logic in chan_sip.c's
handle_invite_replaces() function as well as making use
of ast_do_masquerade() rather than forcing the masquerade
on an ast_read(). The code had several redundant unlocks
that would result in 'freed more times than we've locked!'
errors. I cleaned these up as well as moving all the unlock
logic to the end of the function. This patch should also
resolve the issue people were having with the replacecall
channel never being unlocked with one legged calls.
(closes issue #15151)
Reported by: irroot
Patches:
invite_w_replaces_1.4.diff uploaded by dvossel (license 671)
Tested by: irroot, dvossel
Review: https://reviewboard.asterisk.org/r/371/
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r219324 | mmichelson | 2009-09-17 18:22:01 -0400 (Thu, 17 Sep 2009) | 12 lines
Merged revisions 219320 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r219320 | mmichelson | 2009-09-17 17:20:50 -0500 (Thu, 17 Sep 2009) | 6 lines
Send a 100 Trying response when we detect a spiral.
This was problematic during spiral tests at SIPit...
along with some other things as well.
........
................
r219371 | dvossel | 2009-09-17 18:37:28 -0400 (Thu, 17 Sep 2009) | 9 lines
fixes deadlock when performing directed pickup w Invite/replaces
(closes issue #15340)
Reported by: lmsteffan
Patches:
deadlock.patch uploaded by lmsteffan (license 779)
Tested by: lmsteffan
................
r219412 | tilghman | 2009-09-18 09:54:51 -0400 (Fri, 18 Sep 2009) | 6 lines
Missing value setting line for maxsecs/maxmessage
(closes issue #15696)
Reported by: fhackenberger
Patches:
maxsecs.patch uploaded by fhackenberger (license 592)
................
r219451 | dvossel | 2009-09-18 12:20:41 -0400 (Fri, 18 Sep 2009) | 20 lines
Merged revisions 219450 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r219450 | dvossel | 2009-09-18 11:19:15 -0500 (Fri, 18 Sep 2009) | 14 lines
via-header branches not updated correctly on INVITE
INVITE requests must always contain a new unique branch id. When
a new branch id is created for an INVITE, the dialog's invite_branch
variable must be updated so CANCEL requests use the correct branch id.
(closes issue #15262)
Reported by: maniax
Patches:
asterisk-1.6.1.0-sip-branch.patch uploaded by tweety (license 608)
invite_new_branch_trunk.diff uploaded by dvossel (license 671)
Tested by: maniax, dvossel
........
................
r219520 | dvossel | 2009-09-18 19:20:58 -0400 (Fri, 18 Sep 2009) | 15 lines
Merged revisions 219519 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r219519 | dvossel | 2009-09-18 18:19:50 -0500 (Fri, 18 Sep 2009) | 9 lines
iax2 frame double free
The iax frame's retrans sched id was written over right
before iax2_frame_free was called. In iax2_frame_free that
retrans id is used to delete the sched item. By writing over
the retrans field before the sched item could be deleted, it was
possible for a retransmit to occur on a freed frame.
........
................
r219587 | russell | 2009-09-18 22:59:52 -0400 (Fri, 18 Sep 2009) | 13 lines
Merged revisions 219586 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r219586 | russell | 2009-09-18 21:51:13 -0500 (Fri, 18 Sep 2009) | 6 lines
Make sure the iax_pvt exists before dereferencing it.
This fixes the latest crash posted on issue 15609.
(issue #15609)
........
................
r219654 | tilghman | 2009-09-20 13:55:49 -0400 (Sun, 20 Sep 2009) | 15 lines
Merged revisions 219653 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r219653 | tilghman | 2009-09-20 12:52:05 -0500 (Sun, 20 Sep 2009) | 8 lines
Really stop the stream, when ast_closestream() is called.
(closes issue #15129)
Reported by: bmh
Patches:
20090918__issue15129.diff.txt uploaded by tilghman (license 14)
Review:
https://reviewboard.asterisk.org/r/372/
........
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r219721 | dvossel | 2009-09-21 12:59:05 -0400 (Mon, 21 Sep 2009) | 9 lines
Merged revisions 219720 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r219720 | dvossel | 2009-09-21 11:55:53 -0500 (Mon, 21 Sep 2009) | 3 lines
Reverting merge 219520. This change was not necessary.
........
................
Removed:
team/group/data_api_gsoc2009/contrib/firmware/iax/iaxy.bin
Modified:
team/group/data_api_gsoc2009/ (props changed)
team/group/data_api_gsoc2009/CHANGES
team/group/data_api_gsoc2009/UPGRADE.txt
team/group/data_api_gsoc2009/addons/format_mp3.c
team/group/data_api_gsoc2009/apps/app_chanspy.c
team/group/data_api_gsoc2009/apps/app_directed_pickup.c
team/group/data_api_gsoc2009/apps/app_followme.c
team/group/data_api_gsoc2009/apps/app_meetme.c
team/group/data_api_gsoc2009/apps/app_queue.c
team/group/data_api_gsoc2009/apps/app_voicemail.c
team/group/data_api_gsoc2009/channels/chan_dahdi.c
team/group/data_api_gsoc2009/channels/chan_iax2.c
team/group/data_api_gsoc2009/channels/chan_phone.c
team/group/data_api_gsoc2009/channels/chan_sip.c
team/group/data_api_gsoc2009/channels/sig_analog.c
team/group/data_api_gsoc2009/channels/sig_analog.h
team/group/data_api_gsoc2009/channels/sig_pri.c
team/group/data_api_gsoc2009/channels/sig_pri.h
team/group/data_api_gsoc2009/configs/extensions.conf.sample
team/group/data_api_gsoc2009/configs/voicemail.conf.sample
team/group/data_api_gsoc2009/configure
team/group/data_api_gsoc2009/configure.ac
team/group/data_api_gsoc2009/contrib/scripts/iax-friends.sql
team/group/data_api_gsoc2009/contrib/scripts/live_ast
team/group/data_api_gsoc2009/doc/manager_1_1.txt
team/group/data_api_gsoc2009/include/asterisk/autoconfig.h.in
team/group/data_api_gsoc2009/include/asterisk/cdr.h
team/group/data_api_gsoc2009/include/asterisk/event.h
team/group/data_api_gsoc2009/main/channel.c
team/group/data_api_gsoc2009/main/config.c
team/group/data_api_gsoc2009/main/features.c
team/group/data_api_gsoc2009/main/file.c
team/group/data_api_gsoc2009/main/manager.c
team/group/data_api_gsoc2009/main/pbx.c
team/group/data_api_gsoc2009/res/res_config_odbc.c
team/group/data_api_gsoc2009/res/res_jabber.c
team/group/data_api_gsoc2009/res/res_limit.c
team/group/data_api_gsoc2009/res/res_musiconhold.c
team/group/data_api_gsoc2009/res/res_phoneprov.c
team/group/data_api_gsoc2009/res/res_rtp_asterisk.c
team/group/data_api_gsoc2009/sounds/Makefile
Propchange: team/group/data_api_gsoc2009/
------------------------------------------------------------------------------
Binary property 'branch-1.4-blocked' - no diff available.
Propchange: team/group/data_api_gsoc2009/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.
Propchange: team/group/data_api_gsoc2009/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Tue Sep 22 09:43:42 2009
@@ -1,1 +1,1 @@
-/trunk:1-216929
+/trunk:1-219800
Modified: team/group/data_api_gsoc2009/CHANGES
URL: http://svnview.digium.com/svn/asterisk/team/group/data_api_gsoc2009/CHANGES?view=diff&rev=219804&r1=219803&r2=219804
==============================================================================
--- team/group/data_api_gsoc2009/CHANGES (original)
+++ team/group/data_api_gsoc2009/CHANGES Tue Sep 22 09:43:42 2009
@@ -63,9 +63,11 @@
exit the application.
* The Voicemail application has been improved to automatically ignore messages
that only contain silence.
- * The ChanSpy application now has the 's' option, which makes the application
+ * The ChanSpy application now has the 'S' option, which makes the application
automatically exit once it hits a point where no more channels are available
to spy on.
+ * The ChanSpy application also now has the 'E' option, which spies on a single
+ channel and exits when that channel hangs up.
Dialplan Functions
------------------
Modified: team/group/data_api_gsoc2009/UPGRADE.txt
URL: http://svnview.digium.com/svn/asterisk/team/group/data_api_gsoc2009/UPGRADE.txt?view=diff&rev=219804&r1=219803&r2=219804
==============================================================================
--- team/group/data_api_gsoc2009/UPGRADE.txt (original)
+++ team/group/data_api_gsoc2009/UPGRADE.txt Tue Sep 22 09:43:42 2009
@@ -19,6 +19,12 @@
===========================================================
From 1.6.2 to 1.6.3:
+
+* The firmware for the IAXy has been removed from Asterisk. It can be
+ downloaded from http://downloads.digium.com/pub/iaxy/. To have Asterisk
+ install the firmware into its proper location, place the firmware in the
+ contrib/firmware/iax/ directory in the Asterisk source tree before running
+ "make install".
* There have been some changes to the IAX2 protocol to address the security
concerns documented in the security advisory AST-2009-006. Please see the
Modified: team/group/data_api_gsoc2009/addons/format_mp3.c
URL: http://svnview.digium.com/svn/asterisk/team/group/data_api_gsoc2009/addons/format_mp3.c?view=diff&rev=219804&r1=219803&r2=219804
==============================================================================
--- team/group/data_api_gsoc2009/addons/format_mp3.c (original)
+++ team/group/data_api_gsoc2009/addons/format_mp3.c Tue Sep 22 09:43:42 2009
@@ -200,7 +200,7 @@
if(p->dbuflen) {
for(p->buflen=0; p->buflen < MP3_BUFLEN && p->buflen < p->dbuflen; p->buflen++) {
- s->buf[p->buflen] = p->dbuf[p->buflen+p->dbufoffset];
+ s->buf[p->buflen + AST_FRIENDLY_OFFSET] = p->dbuf[p->buflen+p->dbufoffset];
p->sbufoffset++;
}
p->dbufoffset += p->buflen;
@@ -211,7 +211,7 @@
return NULL;
for(save = p->buflen; p->buflen < MP3_BUFLEN; p->buflen++) {
- s->buf[p->buflen] = p->dbuf[(p->buflen-save)+p->dbufoffset];
+ s->buf[p->buflen + AST_FRIENDLY_OFFSET] = p->dbuf[(p->buflen-save)+p->dbufoffset];
p->sbufoffset++;
}
p->dbufoffset += (MP3_BUFLEN - save);
Modified: team/group/data_api_gsoc2009/apps/app_chanspy.c
URL: http://svnview.digium.com/svn/asterisk/team/group/data_api_gsoc2009/apps/app_chanspy.c?view=diff&rev=219804&r1=219803&r2=219804
==============================================================================
--- team/group/data_api_gsoc2009/apps/app_chanspy.c (original)
+++ team/group/data_api_gsoc2009/apps/app_chanspy.c Tue Sep 22 09:43:42 2009
@@ -71,6 +71,11 @@
<para>Instead of whispering on a single channel barge in on both
channels involved in the call.</para>
</option>
+ <option name="c">
+ <argument name="digit" required="true">
+ <para>Specify a DTMF digit that can be used to spy on the next available channel.</para>
+ </argument>
+ </option>
<option name="d">
<para>Override the typical numeric DTMF functionality and instead
use DTMF to switch between spy modes.</para>
@@ -86,6 +91,15 @@
</enum>
</enumlist>
</option>
+ <option name="e">
+ <argument name="ext" required="true" />
+ <para>Enable <emphasis>enforced</emphasis> mode, so the spying channel can
+ only monitor extensions whose name is in the <replaceable>ext</replaceable> : delimited
+ list.</para>
+ </option>
+ <option name="E">
+ <para>Exit when the spied-on channel hangs up.</para>
+ </option>
<option name="g">
<argument name="grp" required="true">
<para>Only spy on channels in which one or more of the groups
@@ -106,6 +120,9 @@
<argument name="mailbox" />
<argument name="context" />
</option>
+ <option name="o">
+ <para>Only listen to audio coming from this channel.</para>
+ </option>
<option name="q">
<para>Don't play a beep when beginning to spy on a channel, or speak the
selected channel name.</para>
@@ -119,6 +136,9 @@
<para>Skip the playback of the channel type (i.e. SIP, IAX, etc) when
speaking the selected channel name.</para>
</option>
+ <option name="S">
+ <para>Stop when no more channels are left to spy on.</para>
+ </option>
<option name="v">
<argument name="value" />
<para>Adjust the initial volume in the range from <literal>-4</literal>
@@ -132,11 +152,10 @@
<para>Enable <literal>private whisper</literal> mode, so the spying channel can
talk to the spied-on channel but cannot listen to that channel.</para>
</option>
- <option name="o">
- <para>Only listen to audio coming from this channel.</para>
- </option>
- <option name="s">
- <para>Stop when no more channels are left to spy on.</para>
+ <option name="x">
+ <argument name="digit" required="true">
+ <para>Specify a DTMF digit that can be used to exit the application.</para>
+ </argument>
</option>
<option name="X">
<para>Allow the user to exit ChanSpy to a valid single digit
@@ -144,22 +163,6 @@
specified by the <variable>SPY_EXIT_CONTEXT</variable> channel variable. The
name of the last channel that was spied on will be stored
in the <variable>SPY_CHANNEL</variable> variable.</para>
- </option>
- <option name="x">
- <argument name="digit" required="true">
- <para>Specify a DTMF digit that can be used to exit the application.</para>
- </argument>
- </option>
- <option name="c">
- <argument name="digit" required="true">
- <para>Specify a DTMF digit that can be used to spy on the next available channel.</para>
- </argument>
- </option>
- <option name="e">
- <argument name="ext" required="true" />
- <para>Enable <emphasis>enforced</emphasis> mode, so the spying channel can
- only monitor extensions whose name is in the <replaceable>ext</replaceable> : delimited
- list.</para>
</option>
</optionlist>
</parameter>
@@ -205,6 +208,11 @@
<para>Instead of whispering on a single channel barge in on both
channels involved in the call.</para>
</option>
+ <option name="c">
+ <argument name="digit" required="true">
+ <para>Specify a DTMF digit that can be used to spy on the next available channel.</para>
+ </argument>
+ </option>
<option name="d">
<para>Override the typical numeric DTMF functionality and instead
use DTMF to switch between spy modes.</para>
@@ -220,6 +228,15 @@
</enum>
</enumlist>
</option>
+ <option name="e">
+ <argument name="ext" required="true" />
+ <para>Enable <emphasis>enforced</emphasis> mode, so the spying channel can
+ only monitor extensions whose name is in the <replaceable>ext</replaceable> : delimited
+ list.</para>
+ </option>
+ <option name="E">
[... 4065 lines stripped ...]
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