[asterisk-commits] phsultan: branch phsultan/rtmp-support r218090 - in /team/phsultan/rtmp-suppo...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Sep 11 10:40:56 CDT 2009
Author: phsultan
Date: Fri Sep 11 10:40:44 2009
New Revision: 218090
URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=218090
Log:
Merged revisions 214355,214360,214466,214514,214518,214611,214650,214654,214696,214702,214777,214819,214863,214898,214945,215023,215069-215070,215110,215161,215212,215222,215301,215338,215382,215419,215462,215466,215479,215522,215567,215608,215622,215665,215681,215757-215758,215800-215801,215838,215891,215955,216001,216006,216009,216092,216094,216186,216222,216264,216335,216368,216431,216433,216437-216438,216506,216547,216551,216593-216594,216652,216694-216695,216735,216748,216769,216802-216806,216826,216834,216841-216842,216846,216883,216905,216912,216917,216955-216956,216993,217015,217033,217074,217113,217158,217199,217236,217286,217331-217332,217367-217368,217408,217445,217482,217524,217560,217593,217638,217663,217669,217730,217737,217744,217804,217807,217873,217912,217916,217918,217954,217987,217990,218050 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r214355 | jpeeler | 2009-08-27 17:57:47 +0200 (Thu, 27 Aug 2009) | 2 lines
Add forgotten documentation for new channel variables added in 214309.
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r214360 | tilghman | 2009-08-27 18:12:03 +0200 (Thu, 27 Aug 2009) | 10 lines
Merged revisions 214357 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r214357 | tilghman | 2009-08-27 11:03:50 -0500 (Thu, 27 Aug 2009) | 3 lines
Make autoheader descriptions render correctly in our autoconfig.h file.
(Figured out while working with issue #14906)
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r214466 | tilghman | 2009-08-27 19:28:01 +0200 (Thu, 27 Aug 2009) | 9 lines
Merged revisions 214436 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r214436 | tilghman | 2009-08-27 11:53:58 -0500 (Thu, 27 Aug 2009) | 2 lines
One more build system change, to make the descriptions look better, if we have better information.
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r214514 | tilghman | 2009-08-27 23:26:37 +0200 (Thu, 27 Aug 2009) | 7 lines
Ensure that we check for the special value CONFIG_STATUS_FILEINVALID.
(closes issue #15786)
Reported by: a_villacis
Patches:
asterisk-1.6.2.0-beta4-manager-fix-crash-on-include-nonexistent-file.patch uploaded by a villacis (license 660)
(Plus a few of my own, to catch the remaining places within manager.c where it could have been a problem)
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r214518 | tilghman | 2009-08-27 23:46:46 +0200 (Thu, 27 Aug 2009) | 14 lines
Merged revisions 214517 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r214517 | tilghman | 2009-08-27 16:45:34 -0500 (Thu, 27 Aug 2009) | 7 lines
Use autoconf to detect libcurl, as this enables cross-compilation checks, something we didn't allow before.
(closes issue #15714)
Reported by: pprindeville
Patches:
20090813__issue15714.diff.txt uploaded by tilghman (license 14)
Tested by: pprindeville
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r214611 | tilghman | 2009-08-28 18:50:05 +0200 (Fri, 28 Aug 2009) | 6 lines
Remove unnecessary define for Solaris
(closes issue #15358)
Reported by: snuffy
Patches:
bug_res_moh_remove_unneeded_include.diff uploaded by snuffy (license 35)
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r214650 | mmichelson | 2009-08-28 20:41:23 +0200 (Fri, 28 Aug 2009) | 3 lines
Fix some incorrect documentation of sched_thread functions.
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r214654 | rmudgett | 2009-08-28 21:13:53 +0200 (Fri, 28 Aug 2009) | 1 line
Move discardremoteholdretrieval test so it applies only to the specific notification indicator values.
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r214696 | kpfleming | 2009-08-28 22:01:21 +0200 (Fri, 28 Aug 2009) | 9 lines
Ensure that CFLAGS and/or LDFLAGS provided to configure script are preserved.
Cross-compilation environments want to provide 'defaults' for compiler and
linker options, and frequently do this by specifying CFLAGS and LDFLAGS in the
environment or as command-line arguments to the configure script. This patch
modifies the configure script and Makefile to preserve these settings and
ensure they are used in the build process.
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r214702 | tilghman | 2009-08-28 22:14:39 +0200 (Fri, 28 Aug 2009) | 15 lines
Merged revisions 214701 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r214701 | tilghman | 2009-08-28 15:13:32 -0500 (Fri, 28 Aug 2009) | 8 lines
Modify comment to be a bit more accurate.
We have kept this comment around long enough, that it's pretty clear that we're
keeping the code, because changing the code would require a pretty fundamental
architectural shift. We've also taken criticism in some quarters, because it
was believed that it was referring to the code being nasty. No, the code isn't
nasty, just the operation itself is rather odd. Fixed for eternity (probably
not).
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r214777 | russell | 2009-08-29 00:44:44 +0200 (Sat, 29 Aug 2009) | 2 lines
Update configure script so that CONFIG_CFLAGS and CONFIG_LDFLAGS doesn't break the build.
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r214819 | tilghman | 2009-08-30 08:43:04 +0200 (Sun, 30 Aug 2009) | 4 lines
If lua is detected with the lua5.1 prefix (or not), adjust the include path accordingly.
Based upon feedback to a release announcement on the -users list. See
http://lists.digium.com/pipermail/asterisk-users/2009-August/236954.html
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r214863 | tilghman | 2009-08-30 20:37:17 +0200 (Sun, 30 Aug 2009) | 12 lines
Various patches, to enable Asterisk to once again compile on Mac OS X.
One note on defining _POSIX_C_SOURCE: while this feature test macro
works to require certain behaviors on Linux, it works differently on *BSD
platforms to REMOVE certain API calls that are not in the POSIX specification,
such as vasprintf(3). Thus, defining it while depending upon vasprintf (and
other extensions to the POSIX standard) to be defined is a recipe to ensure
that Asterisk is only buildable on Linux.
Hence, this define which was meant to INCREASE portability, effectively
ensures the opposite.
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r214898 | tilghman | 2009-08-31 00:10:35 +0200 (Mon, 31 Aug 2009) | 2 lines
Force Darwin on ppc platforms to compile with a target level that supports aliasing.
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r214945 | tilghman | 2009-08-31 18:18:33 +0200 (Mon, 31 Aug 2009) | 14 lines
Merged revisions 214940 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r214940 | tilghman | 2009-08-31 11:16:52 -0500 (Mon, 31 Aug 2009) | 7 lines
Also unlock the "other" channel, when returning, due to glare.
(closes issue #15787)
Reported by: tim_ringenbach
Patches:
chan_local.diff uploaded by tim ringenbach (license 540)
Tested by: tim_ringenbach
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r215023 | oej | 2009-08-31 20:17:38 +0200 (Mon, 31 Aug 2009) | 2 lines
By copying this code I got bad comments in reviewboard... Better fix the original.
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r215069 | tilghman | 2009-08-31 23:45:00 +0200 (Mon, 31 Aug 2009) | 7 lines
Properly initialize the session to prevent a crash.
(closes issue #15774)
Reported by: lasko
Patches:
20090831__issue15774.diff.txt uploaded by tilghman (license 14)
Tested by: lasko
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r215070 | tilghman | 2009-09-01 00:02:24 +0200 (Tue, 01 Sep 2009) | 2 lines
Fix a trunk compilation warning.
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r215110 | oej | 2009-09-01 16:40:42 +0200 (Tue, 01 Sep 2009) | 2 lines
Removing whitespace that causes red dots in reviewboard
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r215161 | kpfleming | 2009-09-01 21:50:48 +0200 (Tue, 01 Sep 2009) | 3 lines
Ensure that frame dumps of AST_CONTROL_T38_PARAMETERS frames are properly
decoded.
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r215212 | russell | 2009-09-01 22:44:13 +0200 (Tue, 01 Sep 2009) | 45 lines
Fix memory corruption caused by format_mp3.
format_mp3 claimed that it provided AST_FRIENDLY_OFFSET in frames returned by
read(). However, it lied. This means that other parts of the code that
attempted to make use of the offset buffer would end up corrupting the fields
in the ast_filestream structure. This resulted in quite a few crashes due to
unexpected values for fields in ast_filestream.
This patch closes out quite a few bugs. However, some of these bugs have been
open for a while and have been an area where more than one bug has been
discussed. So with that said, anyone that is following one of the issues
closed here, if you still have a problem, please open a new bug report for the
specific problem you are still having. If you do, please ensure that the bug
report is based on the newest version of Asterisk, and that this patch is
applied if format_mp3 is in use. Thanks!
(closes issue #15109)
Reported by: jvandal
Tested by: aragon, russell, zerohalo, marhbere, rgj
(closes issue #14958)
Reported by: aragon
(closes issue #15123)
Reported by: axisinternet
(closes issue #15041)
Reported by: maxnuv
(closes issue #15396)
Reported by: aragon
(closes issue #15195)
Reported by: amorsen
Tested by: amorsen
(closes issue #15781)
Reported by: jensvb
(closes issue #15735)
Reported by: thom4fun
(closes issue #15460)
Reported by: marhbere
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r215222 | tilghman | 2009-09-01 23:19:40 +0200 (Tue, 01 Sep 2009) | 3 lines
Fix register such that lines with a transport string, but without an authuser, parse correctly.
(AST-228)
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r215301 | tilghman | 2009-09-02 01:41:06 +0200 (Wed, 02 Sep 2009) | 8 lines
Add MASTER_CHANNEL() dialplan function, as well as a useful usage.
(closes issue #13140)
Reported by: cpina
Patches:
20090807__issue13140.diff.txt uploaded by tilghman (license 14)
Tested by: lmadsen
Change-type: feature
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r215338 | dhubbard | 2009-09-02 03:16:59 +0200 (Wed, 02 Sep 2009) | 18 lines
Merged revisions 215270 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r215270 | dhubbard | 2009-09-01 18:04:52 -0500 (Tue, 01 Sep 2009) | 12 lines
Use strrchr() so SoftHangup will correctly truncate multi-hyphen channel names
In general channel names are in the form Foo/Bar-Z, but the channel name
could have multiple hyphens and look like Foo/B-a-r-Z. Use strrchr to
truncate the channel name at the last hyphen.
(closes issue #15810)
Reported by: dhubbard
Patches:
dw-softhangup-1.4.patch uploaded by dhubbard (license 733)
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r215382 | oej | 2009-09-02 08:23:05 +0200 (Wed, 02 Sep 2009) | 5 lines
Adding MUTEAUDIO() dialplan function and MuteAudio AMI action (pinepeach)
Review: https://reviewboard.asterisk.org/r/345/
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r215419 | mvanbaak | 2009-09-02 12:50:49 +0200 (Wed, 02 Sep 2009) | 2 lines
Let's compile again on OpenBSD
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r215462 | mvanbaak | 2009-09-02 17:56:46 +0200 (Wed, 02 Sep 2009) | 12 lines
Honor configured parkinglot when parking and retrieving parked calls
Thank oej for pointing out the fact that sip_new did not copy parkinglot from the peer
into the newly created channel.
(closes issue #15538)
Reported by: gracedman
Patches:
2009090100_sipnewparkinglot-161.diff.txt uploaded by mvanbaak (license 7)
With mod by me to also fix callparking as well (this uploaded patch only fixed retrieving a parked call)
Tested by: gracedman, mvanbaak
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r215466 | dvossel | 2009-09-02 18:08:00 +0200 (Wed, 02 Sep 2009) | 16 lines
SIP support for keep-alive event
keep-alive events are used by Sipura/Linksys for NAT keepalive.
There currently don't appear to be any problems with NAT, but
everytime a keep-alive event is received, Asterisk responds with a
"489 Bad event". This error may indicate to a user that NAT
problems exist just because this even is not supported. Now,
rather than respond with an error, the packet is consumed and
a "200 ok" is sent just to indicate we received the packet.
(issue #15084)
Patches:
chan_sip.keepalive.v1.diff uploaded by IgorG (license 20)
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r215479 | mvanbaak | 2009-09-02 18:20:23 +0200 (Wed, 02 Sep 2009) | 3 lines
like in chan_sip's sip_new skinny should copy the configured parkinglot from a line to the newly created channel.
This makes callparking honor the configured parkinglot for skinny lines as well.
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r215522 | dvossel | 2009-09-02 19:26:40 +0200 (Wed, 02 Sep 2009) | 11 lines
SIP uri parsing cleanup
Now, the scheme passed to parse_uri can either be a
single scheme, or a list of schemes ',' delimited.
This gets rid of the whole problem of having to create
two buffers and calling parse_uri twice to check for
separate schemes.
Review: https://reviewboard.asterisk.org/r/343/
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r215567 | tilghman | 2009-09-02 20:37:25 +0200 (Wed, 02 Sep 2009) | 9 lines
Close up to the soft open file limit (same on Linux, but varies drastically on OS X).
Also, a Makefile fix for Darwin (OS X).
(closes issue #14542)
Reported by: jtodd
Patches:
20090901__issue14542.diff.txt uploaded by tilghman (license 14)
Tested by: jtodd, tilghman
Change-type: bugfix
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r215608 | dbailey | 2009-09-02 21:49:43 +0200 (Wed, 02 Sep 2009) | 4 lines
Fix issue where DTMF CID detect was placing channels into signed linear mode
made analog_set_linear_mode return back the mode that was being overwritten
so it could be restored later.
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r215622 | mvanbaak | 2009-09-02 22:21:51 +0200 (Wed, 02 Sep 2009) | 4 lines
- lock channel before looking for a channel variable
- Init the parkings list member of struct parkinglot.
Thanks Sean for the explanation why this should be here.
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r215665 | mvanbaak | 2009-09-02 23:23:17 +0200 (Wed, 02 Sep 2009) | 5 lines
add Parkinglot info to sip show peer <foo> and skinny show line <foo>
If we had this from the start, debugging the 'parking not using configured parkinglot'
bug would have been easier.
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r215681 | dvossel | 2009-09-02 23:39:31 +0200 (Wed, 02 Sep 2009) | 10 lines
port string to int conversion using sscanf
There are several instances where a port is parsed
from a uri or some other source and converted to
an int value using atoi(), if for some reason the
port string is empty, then a standard port is used.
This logic is used over and over, so I created a function
to handle it in a safer way using sscanf().
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r215757 | rmudgett | 2009-09-03 01:25:33 +0200 (Thu, 03 Sep 2009) | 2 lines
Made chan_dahdi able to ignore incoming calls that are not in a MSN list for ISDN PTMP CPE spans.
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r215758 | twilson | 2009-09-03 01:31:04 +0200 (Thu, 03 Sep 2009) | 25 lines
Merged revisions 215682 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r215682 | twilson | 2009-09-02 16:41:22 -0500 (Wed, 02 Sep 2009) | 18 lines
Re-send non-100 provisional responses to prevent cancellation
From section 13.3.1.1 of RFC 3261:
If the UAS desires an extended period of time to answer the INVITE,
it will need to ask for an "extension" in order to prevent proxies
from canceling the transaction. A proxy has the option of canceling
a transaction when there is a gap of 3 minutes between responses in a
transaction. To prevent cancellation, the UAS MUST send a non-100
provisional response at every minute, to handle the possibility of
lost provisional responses.
(closes issue #11157)
Reported by: rjain
Tested by: twilson
Review: https://reviewboard.asterisk.org/r/315/
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r215800 | tilghman | 2009-09-03 05:30:42 +0200 (Thu, 03 Sep 2009) | 5 lines
Revert attempt to standardize with _POSIX_C_SOURCE.
This did not function in the way that was intended, causing more compatibility
issues than it solved. It is best, therefore, that it be simply removed.
(Discussed with kpfleming; agreement to remove was reached.)
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r215801 | tilghman | 2009-09-03 05:43:51 +0200 (Thu, 03 Sep 2009) | 5 lines
Default the callback extension to "s". This is a regression.
(closes issue #15764)
Reported by: elguero
Change-type: bugfix
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r215838 | mvanbaak | 2009-09-03 07:57:23 +0200 (Thu, 03 Sep 2009) | 5 lines
Document that SIPshowpeer and SKINNYshowline now include
the configured parkinglot in their response.
Prodded by snuff-work on #asterisk-dev IRC channel
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r215891 | oej | 2009-09-03 15:02:41 +0200 (Thu, 03 Sep 2009) | 10 lines
Add known internal IP address when autodomain=yes
(closes issue #14573)
Reported by: pj
Patches:
sip-internip-autodomain1.diff uploaded by mnicholson (license 96)
modified by oej
Tested by: pj
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r215955 | dvossel | 2009-09-03 18:31:54 +0200 (Thu, 03 Sep 2009) | 6 lines
Merge code associated with AST-2009-006
(closes issue #12912)
Reported by: rathaus
Tested by: tilghman, russell, dvossel, dbrooks
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r216001 | dvossel | 2009-09-03 20:33:52 +0200 (Thu, 03 Sep 2009) | 12 lines
Blocked revisions 216000 via svnmerge
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r216000 | dvossel | 2009-09-03 13:32:32 -0500 (Thu, 03 Sep 2009) | 7 lines
Merge code associated with AST-2009-006
(closes issue #12912)
Reported by: rathaus
Tested by: tilghman, russell, dvossel, dbrooks
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r216006 | kpfleming | 2009-09-03 20:42:38 +0200 (Thu, 03 Sep 2009) | 14 lines
Document language prompt submission process.
This patch adds a document describing the language prompt submission process,
licensing terms and other issues related to that process. In addition, it
modifies the sound file searching process to support language codes with
any number of suffices (not limited to just "xx" or "xx_YY"), so that prompts
can be named with gender, customer/company, etc. suffices as well.
(closes issue #15771)
Reported by: jtodd
Patches:
language-criteria.txt uploaded by jtodd
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r216009 | russell | 2009-09-03 20:45:54 +0200 (Thu, 03 Sep 2009) | 16 lines
Merged revisions 216008 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r216008 | russell | 2009-09-03 13:44:58 -0500 (Thu, 03 Sep 2009) | 9 lines
Merged revisions 216005 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r216005 | russell | 2009-09-03 13:42:24 -0500 (Thu, 03 Sep 2009) | 2 lines
Add IAX2 security document related to AST-2009-006.
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r216092 | russell | 2009-09-03 21:38:35 +0200 (Thu, 03 Sep 2009) | 16 lines
Merged revisions 216085 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r216085 | russell | 2009-09-03 14:36:46 -0500 (Thu, 03 Sep 2009) | 9 lines
Merged revisions 216080 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r216080 | russell | 2009-09-03 14:35:23 -0500 (Thu, 03 Sep 2009) | 2 lines
Add a note about IAX2 to UPGRADE.txt.
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r216094 | dbailey | 2009-09-03 21:40:37 +0200 (Thu, 03 Sep 2009) | 12 lines
Added detection DTMF CID without polarity change alert.
Added detection of DTMF tone energy levels on FXO channels in chan_dahdi
monitoring loop so DTMF CID can be detected without the need of a polarity
change precursor.
(closes issue #9096)
Reported by: fleed
Patches:
9096-chan_dahdi-trunk.diff uploaded by dbailey (license 819)
Tested by: cyberplant, sum, maturs
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r216186 | rmudgett | 2009-09-03 23:09:46 +0200 (Thu, 03 Sep 2009) | 1 line
Lets try not to use C++ keywords for variable names.
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r216222 | mvanbaak | 2009-09-04 08:08:33 +0200 (Fri, 04 Sep 2009) | 3 lines
make sure 'start' is always initialized.
Makes asterisk compile with --enable-dev-mode
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r216264 | russell | 2009-09-04 12:48:44 +0200 (Fri, 04 Sep 2009) | 16 lines
Merged revisions 216263 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r216263 | russell | 2009-09-04 05:48:00 -0500 (Fri, 04 Sep 2009) | 9 lines
Merged revisions 216262 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r216262 | russell | 2009-09-04 05:47:37 -0500 (Fri, 04 Sep 2009) | 2 lines
Add a plain text version of the IAX2 security document.
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r216335 | oej | 2009-09-04 14:05:46 +0200 (Fri, 04 Sep 2009) | 15 lines
Adding to the janitor list.
For new readers: The janitor list is a list of tasks we need help with in the Asterisk project. Taking up
one of these is often a good way to get into Asterisk development and getting a lot of developers in
the project to be grateful. It's stuff we could spend time on when the bug tracker is empty, when our
employers hasn't filled our task lists and our servers is running bugfree and happily without any issues.
If you want to start working on one of these small projects, feel free to ask for help in the #asterisk-dev
channel on IRC or asterisk-dev mailing list. We'll be more than happy to help you to start and reach
goal.
Thank you for your help.
</end of long commit message>
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r216368 | russell | 2009-09-04 15:14:25 +0200 (Fri, 04 Sep 2009) | 12 lines
Do not treat every SIP peer as if they were configured with insecure=port.
There was a problem in the function responsible for doing peer matching by
IP address and port number such that during the second pass for checking for
a peer configured with insecure=port, it would end up treating every peer as
if it had been configured that way. These changes fix the logic in the peer
IP and port comparison callback to handle insecure=port checking properly.
This problem was introduced when SIP peers were converted to astobj2. Many
thanks to dvossel for noticing this while working on another peer matching
issue.
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r216431 | mvanbaak | 2009-09-04 15:46:59 +0200 (Fri, 04 Sep 2009) | 11 lines
Recorded merge of revisions 216369 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r216369 | mvanbaak | 2009-09-04 15:16:29 +0200 (Fri, 04 Sep 2009) | 4 lines
Make sure 'start' is always initialized.
This is the same as rev 216222 in trunk but 1.4 is affected as well
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r216433 | mvanbaak | 2009-09-04 15:54:25 +0200 (Fri, 04 Sep 2009) | 9 lines
Recorded merge of revisions 216432 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r216432 | mvanbaak | 2009-09-04 15:53:09 +0200 (Fri, 04 Sep 2009) | 2 lines
make chan_sip compile under devmode again
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r216437 | mvanbaak | 2009-09-04 16:00:38 +0200 (Fri, 04 Sep 2009) | 2 lines
make sure canlog is set so we can compile with DEBUG_THREADS enabled on OpenBSD
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r216438 | oej | 2009-09-04 16:02:34 +0200 (Fri, 04 Sep 2009) | 35 lines
Merged revisions 216430 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27 lines
Make apps send PROGRESS control frame for early media and fix too early media issue in SIP
The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI
links *before* any call progress. The SIP channel receives these frames and by default
signals 183 Session progress and starts sending media. This will cause phones to
play silence and ignore the later 180 ringing message. A bad user experience.
The fix is twofold:
- We discovered that asterisk apps that support early media ("noanswer") did not send
any PROGRESS frame to indicate early media. Fixed.
- We introduce a setting in chan_sip so that users can disable any relay of media frames
before the outbound channel actually indicates any sort of call progress.
In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions
of Asterisk, this will be enabled. We don't assume that it will change your Asterisk
phone experience - only for the better.
We encourage third-party application developers to make sure that if they have applications
that wants to send early media, add a PROGRESS control frame transmission to make sure that
all channel drivers actually will start sending early media. This has not been the default
in Asterisk previous to this patch, so if you got inspiration from our code, you need to
update accordingly. Sorry for the trouble and thanks for your support.
This code has been running for a few months in a large scale installation (over 250
servers with PRI and/or BRI links to old PBX systems).
That's no proof that this is an excellent patch, but, well, it's tested :-)
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r216506 | mvanbaak | 2009-09-04 17:05:05 +0200 (Fri, 04 Sep 2009) | 9 lines
Merged revisions 216435 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r216435 | mvanbaak | 2009-09-04 15:56:10 +0200 (Fri, 04 Sep 2009) | 2 lines
make asterisk compile under devmode with DEBUG_THREADS enabled on OpenBSD
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r216547 | tilghman | 2009-09-04 19:31:44 +0200 (Fri, 04 Sep 2009) | 3 lines
Enable turning off the application delimiter warning with the 'dontwarn' option.
Suggested on the -dev list, and implemented in an alternate way by me.
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r216551 | tilghman | 2009-09-04 19:50:21 +0200 (Fri, 04 Sep 2009) | 2 lines
Fix trunk breakage.
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r216593 | seanbright | 2009-09-04 21:29:02 +0200 (Fri, 04 Sep 2009) | 1 line
Use ast_free() instead of free().
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r216594 | dvossel | 2009-09-04 21:32:07 +0200 (Fri, 04 Sep 2009) | 8 lines
sip peer matching by address only with TCP/TLS
This patch removes the contact header matching logic and
adds logic to match all tcp/tls connections by ip only.
Thanks to oej for finding the issue and suggesting solutions.
Review: https://reviewboard.asterisk.org/r/354/
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r216652 | oej | 2009-09-07 13:31:19 +0200 (Mon, 07 Sep 2009) | 2 lines
Simplify the code in this function
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r216694 | oej | 2009-09-07 14:41:08 +0200 (Mon, 07 Sep 2009) | 2 lines
Update sip.conf.sample documentation, reorganize a bit
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r216695 | oej | 2009-09-07 15:06:19 +0200 (Mon, 07 Sep 2009) | 8 lines
If there is no session timer in the INVITE, set it to default value (not unset minimum = -1)
Patch by oej
closes issue #15621
Reported by: fnordian
Tested by: atis
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r216735 | oej | 2009-09-07 16:04:40 +0200 (Mon, 07 Sep 2009) | 2 lines
Fix typo
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r216748 | oej | 2009-09-07 16:21:01 +0200 (Mon, 07 Sep 2009) | 2 lines
Add some doxygen
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r216769 | oej | 2009-09-07 16:54:14 +0200 (Mon, 07 Sep 2009) | 2 lines
Don't send MESSAGE with sendtext() if recepient doesn't allow MESSAGE requests
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r216802 | oej | 2009-09-07 17:47:40 +0200 (Mon, 07 Sep 2009) | 2 lines
Remove unneeded header files (tested on Linux and OS/X)
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r216803 | oej | 2009-09-07 17:48:41 +0200 (Mon, 07 Sep 2009) | 2 lines
After many years, remove VOCAL_DATA_HACK definition
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r216804 | oej | 2009-09-07 18:00:41 +0200 (Mon, 07 Sep 2009) | 2 lines
add doxygen and remove duplicate declaration of variable
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r216805 | oej | 2009-09-07 18:08:08 +0200 (Mon, 07 Sep 2009) | 2 lines
Since it's possible to have more than 999 calls, I'm changing the call counter roof to something higher.
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r216806 | oej | 2009-09-07 18:16:58 +0200 (Mon, 07 Sep 2009) | 2 lines
Doxygen updates
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r216826 | oej | 2009-09-07 18:23:39 +0200 (Mon, 07 Sep 2009) | 2 lines
Move contact_ha to sip_cfg structure
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r216834 | oej | 2009-09-07 18:26:04 +0200 (Mon, 07 Sep 2009) | 2 lines
Move global_regcontext into the sip_cfg structure
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r216841 | oej | 2009-09-07 18:31:36 +0200 (Mon, 07 Sep 2009) | 2 lines
Move capability into sip_cfg. While at it, make sure we reset it at channel reload.
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r216842 | oej | 2009-09-07 18:35:12 +0200 (Mon, 07 Sep 2009) | 2 lines
Make sure we reset global_exclude_static at channel reload
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r216846 | tilghman | 2009-09-07 19:15:37 +0200 (Mon, 07 Sep 2009) | 2 lines
Allow multiple rows to be fetched within the normal mode of operation.
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r216883 | oej | 2009-09-07 20:00:48 +0200 (Mon, 07 Sep 2009) | 6 lines
Clean up the "offered_media" code
- Add variable for number of known media streams instead of hardcoding in definition of sip_pvt
- Rename "text" to "codecs" - beacuse it's what it is
- Add documentation for future developers so that we make sure that we define new sdp media types
for SRTP-variants
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r216905 | oej | 2009-09-07 20:24:04 +0200 (Mon, 07 Sep 2009) | 5 lines
- Doxygen additions
- Remove unused string in sip_registry -- "random"
- Someone added a function in the middle of all forward declarations... Weird. Moved it out of that
section.
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r216912 | oej | 2009-09-07 20:26:37 +0200 (Mon, 07 Sep 2009) | 2 lines
Move "deprecated_username" to a flag like the others - unsigned int blah:1
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r216917 | oej | 2009-09-07 20:29:45 +0200 (Mon, 07 Sep 2009) | 5 lines
Moving another function declared in the middle of forward declarations.
Please follow the structure of the source code, thanks. Chan_sip is messy enough as it is :-)
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r216955 | oej | 2009-09-07 22:19:37 +0200 (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 22:23:19 +0200 (Mon, 07 Sep 2009) | 2 lines
Fixing formatting
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r216993 | dvossel | 2009-09-08 16:26:30 +0200 (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 17:23:04 +0200 (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 17:30:18 +0200 (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 18:37:28 +0200 (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 20:06:57 +0200 (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 22:06:15 +0200 (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 22:28:41 +0200 (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 23:17:16 +0200 (Tue, 08 Sep 2009) | 1 line
Remove duplicate entry in the sig_pri_pri private pointer array.
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r217286 | seanbright | 2009-09-09 00:17:08 +0200 (Wed, 09 Sep 2009) | 4 lines
Fix compilation of app_meetme.
Reported by ebroad in #asterisk-bugs
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r217331 | rmudgett | 2009-09-09 01:31:27 +0200 (Wed, 09 Sep 2009) | 1 line
Miscellaneous minor code cleanup in mkintf().
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r217332 | rmudgett | 2009-09-09 01:37:57 +0200 (Wed, 09 Sep 2009) | 1 line
Fix memory leak of sig_xxx private structures.
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r217367 | oej | 2009-09-09 12:38:45 +0200 (Wed, 09 Sep 2009) | 2 lines
Formatting and doxygen updates
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r217368 | oej | 2009-09-09 12:39:43 +0200 (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 14:11:12 +0200 (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 20:52:48 +0200 (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 22:09:31 +0200 (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 23:48:04 +0200 (Wed, 09 Sep 2009) | 1 line
ast_log replaced for ast_verbose in MFCR2 event notifications
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r217560 | rmudgett | 2009-09-10 02:35:30 +0200 (Thu, 10 Sep 2009) | 1 line
Fix available() for SS7, MFC/R2, and pseudo channels.
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r217593 | oej | 2009-09-10 14:06:55 +0200 (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 20:17:14 +0200 (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 20:29:21 +0200 (Thu, 10 Sep 2009) | 2 lines
Don't assign UINT_MAX to an INT.
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r217669 | oej | 2009-09-10 21:09:02 +0200 (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 21:39:41 +0200 (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 21:55:16 +0200 (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 22:18:30 +0200 (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 22:52:57 +0200 (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 23:07:47 +0200 (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 | 2009-09-11 00:11:17 +0200 (Fri, 11 Sep 2009) | 1 line
Miscellaneous minor changes.
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r217912 | rmudgett | 2009-09-11 00:31:12 +0200 (Fri, 11 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 | 2009-09-11 01:12:16 +0200 (Fri, 11 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 | 2009-09-11 01:16:24 +0200 (Fri, 11 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 | 2009-09-11 01:29:14 +0200 (Fri, 11 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 | 2009-09-11 01:49:09 +0200 (Fri, 11 Sep 2009) | 2 lines
Cleanup approach in 217804 and don't reach inside the sig_pvt.
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