[svn-commits] phsultan: branch phsultan/rtmp-support r209398 - in /team/phsultan/rtmp-suppo...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Jul 28 07:36:45 CDT 2009
Author: phsultan
Date: Tue Jul 28 07:36:37 2009
New Revision: 209398
URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=209398
Log:
Merged revisions 206808,206868,206873,206877,206939,206998,207029,207034,207093,207095,207156,207224-207225,207255,207285,207317-207318,207361,207424,207484,207522,207551,207599,207680,207723,207854,207902,207925,207934,207946,207950,208017-208018,208052,208113,208151,208155,208193,208229,208263,208267,208314,208383,208388,208464,208504,208542,208548,208588,208593,208630,208693,208706,208709,208749,208813,208848,208886,208924,208991,209056,209098,209132,209197,209235,209256,209279,209317,209331 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r206808 | tilghman | 2009-07-16 18:51:05 +0200 (Thu, 16 Jul 2009) | 13 lines
Merged revisions 206807 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r206807 | tilghman | 2009-07-16 11:27:35 -0500 (Thu, 16 Jul 2009) | 6 lines
Fix a memory leak.
(closes issue #15517)
Reported by: adomjan
Patches:
func_realtime.c-ast_variable_destroy.diff uploaded by adomjan (license 487)
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r206868 | dvossel | 2009-07-16 23:25:22 +0200 (Thu, 16 Jul 2009) | 14 lines
Merged revisions 206867 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r206867 | dvossel | 2009-07-16 16:24:16 -0500 (Thu, 16 Jul 2009) | 8 lines
avoid segfault caused by user error
If the CALLERPRES() dialplan function is set to nothing,
a segfault occurs. This is user error to begin with, but
I'd rather see a cli warning message than have Asterisk
crash on me.
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r206873 | dvossel | 2009-07-16 23:33:51 +0200 (Thu, 16 Jul 2009) | 12 lines
Merged revisions 206872 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r206872 | dvossel | 2009-07-16 16:33:19 -0500 (Thu, 16 Jul 2009) | 6 lines
error in iax.conf related IP-based access control
(closes issue #15518)
Reported by: pkempgen
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r206877 | dvossel | 2009-07-16 23:45:14 +0200 (Thu, 16 Jul 2009) | 6 lines
TIMEOUT(absolute) returned negative value.
(closes issue #15513)
Reported by: ys
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r206939 | dvossel | 2009-07-17 18:13:22 +0200 (Fri, 17 Jul 2009) | 20 lines
Merged revisions 206938 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r206938 | dvossel | 2009-07-17 11:05:06 -0500 (Fri, 17 Jul 2009) | 14 lines
SIP incorrect From: header information when callpres is prohib
Some ITSP make use of the "Anonymous" display name to detect a
requirement to withhold caller id across the PSTN. This does
not work if the display name is "Unknown".
(closes issue #14465)
Reported by: Nick_Lewis
Patches:
chan_sip.c-callerpres.patch uploaded by Nick (license 657)
chan_sip.c-callerpres_trunk.patch uploaded by dvossel (license 671)
Tested by: Nick_Lewis, dvossel
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r206998 | jpeeler | 2009-07-17 19:02:44 +0200 (Fri, 17 Jul 2009) | 14 lines
Fix segfault in sig_analog when using callwaiting, respect callwaiting options
Sig_analog handles allocating the sub channel for callwaiting, so no longer try
to do it in chan_dahdi. Modified analog_alloc_sub to only mark the sub as
allocated upon success of the alloc_sub callback, which was responsible for the
segfault. Also, the callwaiting and callwaitingcallerid options were being
unconditionally set to true. Now, the options are properly set from
chan_dahdi.conf.
(closes issue #15508)
Reported by: elguero
Tested by: elguero
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r207029 | dvossel | 2009-07-17 19:51:44 +0200 (Fri, 17 Jul 2009) | 6 lines
sip option flags handled incorrectly
(closes issue #15376)
Reported by: Takehiko Ooshima
Tested by: dvossel, Takehiko_Ooshima
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r207034 | dvossel | 2009-07-17 20:01:04 +0200 (Fri, 17 Jul 2009) | 10 lines
Blocked revisions 207033 via svnmerge
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r207033 | dvossel | 2009-07-17 13:00:38 -0500 (Fri, 17 Jul 2009) | 4 lines
sip option flags handled incorrectly
(issue #15376)
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r207093 | jpeeler | 2009-07-17 21:14:02 +0200 (Fri, 17 Jul 2009) | 16 lines
Blocked revisions 207092 via svnmerge
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r207092 | jpeeler | 2009-07-17 14:13:27 -0500 (Fri, 17 Jul 2009) | 11 lines
Enhance configuration option for overlapdial allowing direction choice
Previously overlap dialing could only be turned on or off for both incoming and
outgoing calls. New parameters incoming, outgoing, and both have been added to
allow further control. There is no change in default behavior with these new
options and allows in band DTMF to be accepted in one direction if required.
(closes issue #14471)
Reported by: eboscani
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r207095 | jpeeler | 2009-07-17 21:16:35 +0200 (Fri, 17 Jul 2009) | 2 lines
Update some missing allowed options for overlapdial
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r207156 | jpeeler | 2009-07-17 21:37:38 +0200 (Fri, 17 Jul 2009) | 14 lines
Merged revisions 207155 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r207155 | jpeeler | 2009-07-17 14:36:19 -0500 (Fri, 17 Jul 2009) | 7 lines
Fix format specifier to print out an unsigned long long.
Yep, it's even ifdefed out code. But it made it to the RR list...
(closes issue #14726)
Reported by: lmadsen
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r207224 | tilghman | 2009-07-18 00:04:43 +0200 (Sat, 18 Jul 2009) | 2 lines
Document the "flag" field in the voicemessages table.
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r207225 | dvossel | 2009-07-18 00:07:36 +0200 (Sat, 18 Jul 2009) | 10 lines
fixes an error in r203638 CEL commit
(closes issue #15525)
Reported by: elguero
Patches:
iax2-double-unlock.patch uploaded by elguero (license 37)
15525.diff uploaded by dvossel (license 671)
Tested by: dvossel
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r207255 | tilghman | 2009-07-18 00:29:50 +0200 (Sat, 18 Jul 2009) | 2 lines
Add flag here, too (as requested by jsmith)
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r207285 | rmudgett | 2009-07-18 03:31:53 +0200 (Sat, 18 Jul 2009) | 72 lines
Recorded merge of revisions 145293,158010 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r145293 | rmudgett | 2008-09-30 18:55:24 -0500 (Tue, 30 Sep 2008) | 54 lines
channels/chan_misdn.c
channels/misdn/isdn_lib.c
* Miscellaneous other fixes from trunk to make merging easier later.
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r145200 | rmudgett | 2008-09-30 16:00:54 -0500 (Tue, 30 Sep 2008) | 7 lines
* Miscellaneous formatting changes to make v1.4 and trunk
more merge compatible in the mISDN area.
channels/chan_misdn.c
* Eliminated redundant code in cb_events() EVENT_SETUP
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r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008) | 9 lines
improved helptext of misdn_set_opt.
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r142181 | rmudgett | 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) | 1 line
Cleaned up comment
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r138738 | rmudgett | 2008-08-18 16:07:28 -0500 (Mon, 18 Aug 2008) | 30 lines
channels/chan_misdn.c
* Made bearer2str() use allowed_bearers_array[]
* Made use the causes.h defines instead of hardcoded numbers.
* Made use Asterisk presentation indicator values if either of the
mISDN presentation or screen options are negative.
* Updated the misdn_set_opt application option descriptions.
* Renamed the awkward Caller ID presentation misdn_set_opt
application option value not_screened to restricted.
Deprecated the not_screened option value.
channels/misdn/isdn_lib.c
* Made use the causes.h defines instead of hardcoded numbers.
* Fixed some spelling errors and typos.
* Added all defined facility code strings to fac2str().
channels/misdn/isdn_lib.h
* Added doxygen comments to struct misdn_bchannel.
channels/misdn/isdn_lib_intern.h
* Added doxygen comments to struct misdn_stack.
channels/misdn_config.c
configs/misdn.conf.sample
* Updated the mISDN presentation and screen parameter descriptions.
doc/misdn.txt (doc/tex/misdn.tex)
* Updated the misdn_set_opt application option descriptions.
* Fixed some spelling errors and typos.
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r158010 | rmudgett | 2008-11-19 19:46:09 -0600 (Wed, 19 Nov 2008) | 9 lines
Merged revision 157977 from
https://origsvn.digium.com/svn/asterisk/team/group/issue8824
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Fixes JIRA ABE-1726
The dial extension could be empty if you are using MISDN_KEYPAD
to control ISDN provider features.
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r207317 | tilghman | 2009-07-18 06:16:44 +0200 (Sat, 18 Jul 2009) | 3 lines
Flag field in wrong position.
Reported by "Hoggins!" on asterisk-dev list.
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r207318 | rmudgett | 2009-07-18 06:17:01 +0200 (Sat, 18 Jul 2009) | 26 lines
Merged 207316 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...
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r207316 | rmudgett | 2009-07-17 23:05:05 -0500 (Fri, 17 Jul 2009) | 20 lines
Fixed incoming calls being matched to MSNs without type-of-number prefix added.
For an incoming ISDN call the dialed.number is incorrectly matched against
the configured MSNs in misdn.conf. The numbers passed to the dialplan
include the configured prefix for the dialed.number_type, whereas the
check against the configured MSNs (to decide if the call is accepted at
all), is executed without the configured prefix.
e.g., dialed.number = 241168020, TON = national, configured national
prefix is "0". (This is the TON which is used by ISDN providers in the
Netherlands.)
In chan_misdn.c:cb_events() in case EVENT_SETUP the call to
misdn_cfg_is_msn_valid() uses the unnormalized number 241168020, but 57
lines later the call to read_config() adds the prefix, and the
dialed.number is now 0241168020, which is then used in the dialplan.
misdn_cfg_is_msn_valid() must use the normalized number, too.
JIRA ABE-1912
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r207361 | russell | 2009-07-20 18:36:15 +0200 (Mon, 20 Jul 2009) | 16 lines
Merged revisions 207360 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r207360 | russell | 2009-07-20 11:26:24 -0500 (Mon, 20 Jul 2009) | 9 lines
Only do the chan->fdno check in ast_read() in a developer build.
I changed this check to only happen in a dev-mode build. I also added a
comment explaining what is going on. I also made it so that detection of
this situation does not affect ast_read() operation.
(closes issue #14723)
Reported by: seadweller
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r207424 | mmichelson | 2009-07-20 21:48:12 +0200 (Mon, 20 Jul 2009) | 39 lines
Merged revisions 207423 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r207423 | mmichelson | 2009-07-20 14:39:59 -0500 (Mon, 20 Jul 2009) | 33 lines
Answer video SDP offers properly when videosupport is not enabled.
Copied from Review board:
In issue 12434, the reporter describes a situation in which audio and video
is offered on the call, but because videosupport is disabled in sip.conf,
Asterisk gives no response at all to the video offer. According to RFC 3264,
all media offers should have a corresponding answer. For offers we do not
intend to actually reply to with meaningful values, we should still reply
with the port for the media stream set to 0.
In this patch, we take note of what types of media have been offered and
save the information on the sip_pvt. The SDP in the response will take into
account whether media was offered. If we are not otherwise going to answer
a media offer, we will insert an appropriate m= line with the port set to 0.
It is important to note that this patch is pretty much a bandage being
applied to a broken bone. The patch *only* helps for situations where video
is offered but videosupport is disabled and when udptl_pt is disabled but
T.38 is offered. Asterisk is not guaranteed to respond to every media offer.
Notable cases are when multiple streams of the same type are offered.
The 2 media stream limit is still present with this patch, too.
In trunk and the 1.6.X branches, things will be a bit different since Asterisk
also supports text in SDPs as well.
(closes issue #12434)
Reported by: mnnojd
Review: https://reviewboard.asterisk.org/r/311
Review: https://reviewboard.asterisk.org/r/313
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r207484 | dvossel | 2009-07-20 22:45:26 +0200 (Mon, 20 Jul 2009) | 16 lines
reg->username is parsed only once on sip reload
The registration string can contain an expanded user portion of the
form user at domain. This expanded user portion was stored in
reg->username and parsed each time there is a registration refresh.
Now, the domain portion of the user is parsed and stored separately
in the regdomain field.
(closes issue #14331)
Reported by: Nick_Lewis
Patches:
chan_sip.c.domainparse3.patch uploaded by Nick (license 657)
Tested by: Nick_Lewis, dvossel
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r207522 | mmichelson | 2009-07-21 00:13:34 +0200 (Tue, 21 Jul 2009) | 7 lines
Initialize connected line instance when doing a directed pickup.
This helps to prevent a crash which may occur due to our freeing
garbage due to a struct being uninitialized.
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r207551 | mmichelson | 2009-07-21 01:08:56 +0200 (Tue, 21 Jul 2009) | 3 lines
Okay, that didn't fix the crash. It didn't really do anything useful.
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r207599 | jpeeler | 2009-07-21 01:31:36 +0200 (Tue, 21 Jul 2009) | 16 lines
Blocked revisions 207573 via svnmerge
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r207573 | jpeeler | 2009-07-20 18:23:18 -0500 (Mon, 20 Jul 2009) | 10 lines
Wait for wink before dialing when using E&M wink signaling
This patch adds a new dahdi_wait function to specifically wait for the wink
event. If the wink is not eventually received the channel is hung up.
(closes issue #14434)
Reported by: araasch
Patches:
emwinkmod uploaded by araasch (license 693)
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r207680 | kpfleming | 2009-07-21 15:28:04 +0200 (Tue, 21 Jul 2009) | 18 lines
Merged revisions 207647 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r207647 | kpfleming | 2009-07-21 08:04:44 -0500 (Tue, 21 Jul 2009) | 12 lines
Ensure that user-provided CFLAGS and LDFLAGS are honored.
This commit changes the build system so that user-provided flags (in ASTCFLAGS
and ASTLDFLAGS) are supplied to the compiler/linker *after* all flags provided
by the build system itself, so that the user can effectively override the
build system's flags if desired. In addition, ASTCFLAGS and ASTLDFLAGS can now
be provided *either* in the environment before running 'make', or as variable
assignments on the 'make' command line. As a result, the use of COPTS and LDOPTS
is no longer necessary, so they are no longer documented, but are still supported
so as not to break existing build systems that supply them when building Asterisk.
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r207723 | mmichelson | 2009-07-21 16:29:40 +0200 (Tue, 21 Jul 2009) | 11 lines
Merged revisions 207714 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r207714 | mmichelson | 2009-07-21 09:26:00 -0500 (Tue, 21 Jul 2009) | 5 lines
Document default timeout for AMI originations.
AST-224
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r207854 | jpeeler | 2009-07-21 22:26:02 +0200 (Tue, 21 Jul 2009) | 16 lines
Merged revisions 207827 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r207827 | jpeeler | 2009-07-21 15:16:55 -0500 (Tue, 21 Jul 2009) | 9 lines
Wait for wink before dialing when using E&M wink signaling
There was already code for other signaling types in dahdi_handle_event to
handle dialing if a dial operation dial string was present. Simply add
SIG_EMWINK to the list.
(closes issue #14434)
Reported by: araasch
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r207902 | jpeeler | 2009-07-22 00:02:25 +0200 (Wed, 22 Jul 2009) | 2 lines
Fix my_is_off_hook to check rxbits only for FXS signaling
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r207925 | russell | 2009-07-22 00:22:18 +0200 (Wed, 22 Jul 2009) | 4 lines
Note that we use tabs instead of spaces for indentation.
I'm surprised this was never actually in here...
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r207934 | jpeeler | 2009-07-22 00:24:56 +0200 (Wed, 22 Jul 2009) | 1 line
whitespace fix only
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r207946 | tilghman | 2009-07-22 00:45:32 +0200 (Wed, 22 Jul 2009) | 15 lines
Merged revisions 207945 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r207945 | tilghman | 2009-07-21 17:38:54 -0500 (Tue, 21 Jul 2009) | 8 lines
Force an error if a blank is passed to QUOTE (because the documentation states the argument is not optional).
This change makes URIENCODE and QUOTE behave similarly, since the documentation
states that the argument is not optional, for both.
(closes issue #15439)
Reported by: pkempgen
Patches:
20090706__issue15439.diff.txt uploaded by tilghman (license 14)
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r207950 | jpeeler | 2009-07-22 00:51:47 +0200 (Wed, 22 Jul 2009) | 7 lines
Do not dial digits when none were specified for sig_pri based calls
(closes issue #15524)
Reported by: elguero
Patches:
pri-sig-no-dest-set.patch uploaded by elguero (license 37)
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r208017 | mmichelson | 2009-07-22 16:35:01 +0200 (Wed, 22 Jul 2009) | 12 lines
Fix the crash in directed pickups. For real this time.
A shallow pointer copy was causing an ast_party_connected_line
structure to be freed multiple times, thus causing a crash.
(closes issue #15441)
Reported by: lmsteffan
Patches:
15441.patch uploaded by mmichelson (license 60)
Tested by: lmsteffan
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r208018 | russell | 2009-07-22 16:35:49 +0200 (Wed, 22 Jul 2009) | 2 lines
Remove trailing whitespace.
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r208052 | tilghman | 2009-07-22 18:49:42 +0200 (Wed, 22 Jul 2009) | 7 lines
Clarify documentation on 'realtime update2' to show more than one condition.
(closes issue #15357)
Reported by: snuffy
Patches:
bug_fix_doc_update2.diff uploaded by snuffy (license 35)
(slightly modified by me)
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r208113 | qwell | 2009-07-22 23:43:57 +0200 (Wed, 22 Jul 2009) | 9 lines
Restore an int declaration on PPC platforms.
This x is one crafty little bugger...
It was used for 2 different things (one of which was only done on PPC) in 1.4.
One of the uses were removed in trunk, and with it went the declaration.
(closes issue #14038)
Reported by: ffloimair
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r208151 | tilghman | 2009-07-23 00:35:57 +0200 (Thu, 23 Jul 2009) | 11 lines
Merged revisions 208083 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r208083 | tilghman | 2009-07-22 15:23:53 -0500 (Wed, 22 Jul 2009) | 4 lines
Export symbols for functions included in our compatibility headers.
(closes issue #15556)
Reported by: smw1218
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r208155 | jpeeler | 2009-07-23 00:42:33 +0200 (Thu, 23 Jul 2009) | 5 lines
Reset the fax buffers back to default settings regardless of signaling in use -
Pointed out by Matt F.
Also in the case of not using a signaling module, set the law back to the
default as well.
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r208193 | russell | 2009-07-23 03:31:18 +0200 (Thu, 23 Jul 2009) | 2 lines
Resolve compiler warning on mac.
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r208229 | mmichelson | 2009-07-23 16:46:53 +0200 (Thu, 23 Jul 2009) | 8 lines
Fix potential crash if p->owner is NULL.
Problem was observed when a call-forwarding loop was accidentally
configured.
ABE-1906
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r208263 | mmichelson | 2009-07-23 17:46:34 +0200 (Thu, 23 Jul 2009) | 15 lines
Merged revisions 208262 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r208262 | mmichelson | 2009-07-23 10:43:07 -0500 (Thu, 23 Jul 2009) | 8 lines
Properly handle 183 responses which do not contain an SDP.
(closes issue #15442)
Reported by: ffloimair
Patches:
15442.patch uploaded by mmichelson (license 60)
Tested by: tkarl, ffloimair
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r208267 | jpeeler | 2009-07-23 17:59:44 +0200 (Thu, 23 Jul 2009) | 13 lines
Fix sending of interface identifier unconditionally in sig_pri
The wrong logic was being used in chan_dahdi to convert a sig_pri_chan
to the proper libpri channel number. The most significant bit must only
be set only when trunk groups are being used.
(closes issue #15452)
Reported by: alecdavis
Patches:
bug15452.patch uploaded by jpeeler (license 325)
Tested by: alecdavis
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r208314 | mmichelson | 2009-07-23 18:29:37 +0200 (Thu, 23 Jul 2009) | 9 lines
Merged revisions 208312 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r208312 | mmichelson | 2009-07-23 11:29:18 -0500 (Thu, 23 Jul 2009) | 3 lines
Remove inaccurate XXX comment.
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r208383 | jpeeler | 2009-07-23 21:21:50 +0200 (Thu, 23 Jul 2009) | 12 lines
Merged revisions 208380 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r208380 | jpeeler | 2009-07-23 14:19:53 -0500 (Thu, 23 Jul 2009) | 6 lines
Only set the priindication setting when not performing a reload
(closes issue #14696)
Reported by: fdecher
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r208388 | mmichelson | 2009-07-23 21:34:49 +0200 (Thu, 23 Jul 2009) | 24 lines
Merged revisions 208386 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r208386 | mmichelson | 2009-07-23 14:24:21 -0500 (Thu, 23 Jul 2009) | 17 lines
Fix a problem where a 491 response could be sent out of dialog.
This generalizes the fix for issue 13849. The initial fix corrected the
problem that Asterisk would reply with a 491 if a reinvite were received
from an endpoint and we had not yet received an ACK from that endpoint
for the initial INVITE it had sent us. This expansion also allows Asterisk
to appropriately handle an INVITE with authorization credentials if Asterisk
had not received an ACK from the previous transaction in which Asterisk had
responded to an unauthorized INVITE with a 407.
(closes issue #14239)
Reported by: klaus3000
Patches:
14239.patch uploaded by mmichelson (license 60)
Tested by: klaus3000
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r208464 | kpfleming | 2009-07-23 23:57:24 +0200 (Thu, 23 Jul 2009) | 46 lines
Rework of T.38 negotiation and UDPTL API to address interoperability problems
Over the past couple of months, a number of issues with Asterisk
negotiating (and successfully completing) T.38 sessions with various
endpoints have been found. This patch attempts to address many of
them, primarily focused around ensuring that the endpoints'
MaxDatagram size is honored, and in addition by ensuring that T.38
session parameter negotiation is performed correctly according to the
ITU T.38 Recommendation.
The major changes here are:
1) T.38 applications in Asterisk (app_fax) only generate/receive IFP
packets, they do not ever work with UDPTL packets. As a result of
this, they cannot be allowed to generate packets that would overflow
the other endpoints' MaxDatagram size after the UDPTL stack adds any
error correction information. With this patch, the application is told
the maximum *IFP* size it can generate, based on a calculation using
the far end MaxDatagram size and the active error correction mode on
the T.38 session. The same is true for sending *our* MaxDatagram size
to the remote endpoint; it is computed from the value that the
application says it can accept (for a single IFP packet) combined with
the active error correction mode.
2) All treatment of T.38 session parameters as 'capabilities' in
chan_sip has been removed; these parameters are not at all like
audio/video stream capabilities. There are strict rules to follow for
computing an answer to a T.38 offer, and chan_sip now follows those
rules, using the desired parameters from the application (or channel)
that wants to accept the T.38 negotiation.
3) chan_sip now stores and forwards ast_control_t38_parameters
structures for tracking 'our' and 'their' T.38 session parameters;
this greatly simplifies negotiation, especially for pass-through
calls.
4) Since T.38 negotiation without specifying parameters or receiving
the final negotiated parameters is not very worthwhile, the
AST_CONTROL_T38 control frame has been removed. A note has been added
to UPGRADE.txt about this removal, since any out-of-tree applications
that use it will no longer function properly until they are upgraded
to use AST_CONTROL_T38_PARAMETERS.
Review: https://reviewboard.asterisk.org/r/310/
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r208504 | kpfleming | 2009-07-24 00:32:52 +0200 (Fri, 24 Jul 2009) | 1 line
T.38 change note is not necessary in this branch
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r208542 | mvanbaak | 2009-07-24 16:35:49 +0200 (Fri, 24 Jul 2009) | 13 lines
use aptitude for debian based systems
The function to check wether we need to install packages was using
dpkg-query which was gives wrong output on Debian 5
Also, the apt-get has been replaced with aptitude because aptitude
is now the preferred way to handle packages on Debian
(closes issue #15570)
Reported by: mvanbaak
Patches:
2009072400_installprereq-aptitude.diff uploaded by mvanbaak (license 7)
................
r208548 | kpfleming | 2009-07-24 17:02:53 +0200 (Fri, 24 Jul 2009) | 8 lines
Resolve a T.38 negotiation issue left over from the udptl-updates merge.
The udptl-updates branch that was merged yesterday failed to properly send back
T.38 SDP responses with the correct error correction mode, if the incoming SDP
from the other end caused us to change error correction modes. This patch
corrects that situation.
................
r208588 | mmichelson | 2009-07-24 20:31:04 +0200 (Fri, 24 Jul 2009) | 16 lines
Merged revisions 208587 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r208587 | mmichelson | 2009-07-24 13:26:50 -0500 (Fri, 24 Jul 2009) | 10 lines
Only send a BYE when hanging up a channel that is up.
For cases where Asterisk sends an INVITE and receives a non 2XX final
response, Asterisk would follow the INVITE transaction by immediately
sending a BYE, which was unnecessary.
(closes issue #14575)
Reported by: chris-mac
........
................
r208593 | russell | 2009-07-24 20:42:32 +0200 (Fri, 24 Jul 2009) | 14 lines
Merged revisions 208592 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r208592 | russell | 2009-07-24 13:38:24 -0500 (Fri, 24 Jul 2009) | 7 lines
Do not log an ERROR if autoservice_stop() returns -1.
This does not indicate an error. A return of -1 just means that the channel
has been hung up.
(reported in #asterisk-dev)
........
................
r208630 | mmichelson | 2009-07-24 21:26:26 +0200 (Fri, 24 Jul 2009) | 21 lines
Blocked revisions 208622 via svnmerge
........
r208622 | mmichelson | 2009-07-24 14:24:28 -0500 (Fri, 24 Jul 2009) | 16 lines
Don't impose an arbitrary limit on member lines in queues.conf
I know what some of you are thinking: "UGH! Mark, why are you using
ast_strdup and ast_free for the string when you can just use ast_strdupa
and let the memory free itself?! Have the bats been chewing on your brain
again?"
Based on past experiences, I don't like using ast_strdupa inside a loop.
It's a good way to potentially exhaust stack space. Also, since this only
happens when reloading queues, I don't think that heap allocations and
frees are going to be a huge problem.
(closes issue #15559)
Reported by: amorsen
........
................
r208693 | russell | 2009-07-24 22:25:23 +0200 (Fri, 24 Jul 2009) | 2 lines
Don't log a warning for something that does not affect operation.
................
r208706 | russell | 2009-07-24 22:54:37 +0200 (Fri, 24 Jul 2009) | 6 lines
Note that "reload" needs to be added back.
I keep getting annoyed at having to type "module reload" to reload everything,
so I'm adding a note that we need to add "reload" back. "module reload" doesn't
really make sense as the command to reload everything, including the core.
................
r208709 | russell | 2009-07-24 23:12:43 +0200 (Fri, 24 Jul 2009) | 2 lines
Remove trailing whitespace.
................
r208749 | jpeeler | 2009-07-25 08:23:18 +0200 (Sat, 25 Jul 2009) | 13 lines
Merged revisions 208746 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r208746 | jpeeler | 2009-07-25 01:19:50 -0500 (Sat, 25 Jul 2009) | 7 lines
Fix compiling under dev-mode with gcc 4.4.0.
Mostly trivial changes, but I did not know of any other way to fix the
"dereferencing type-punned pointer will break strict-aliasing rules" error
without creating a tmp variable in chan_skinny.
........
................
r208813 | mvanbaak | 2009-07-25 14:03:25 +0200 (Sat, 25 Jul 2009) | 10 lines
add default alias reload to run module reload.
Requiring 'module reload' to reload everything, including
core etc makes russell very unhappy.
The default configuration already loads the 'friendly' aliases template.
Added 'reload=module reload' to that template.
Also removed the comment in main/cli.c that reload should come back.
................
r208848 | mvanbaak | 2009-07-25 14:28:38 +0200 (Sat, 25 Jul 2009) | 2 lines
libxml2-dev is needed as well by default.
................
r208886 | mvanbaak | 2009-07-26 16:00:52 +0200 (Sun, 26 Jul 2009) | 2 lines
add OpenBSD to the install_prereq script
................
r208924 | jpeeler | 2009-07-27 03:20:37 +0200 (Mon, 27 Jul 2009) | 9 lines
Merged revisions 208923 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r208923 | jpeeler | 2009-07-26 20:18:31 -0500 (Sun, 26 Jul 2009) | 2 lines
Fix logic errors from 208746
........
................
r208991 | mvanbaak | 2009-07-27 11:56:49 +0200 (Mon, 27 Jul 2009) | 11 lines
Blocked revisions 208990 via svnmerge
........
r208990 | mvanbaak | 2009-07-27 11:56:13 +0200 (Mon, 27 Jul 2009) | 5 lines
backport rev 205532 from trunk:
pthread_self returns a pthread_t which is not an unsigned int on all
pthread implementations. Casting it to an unsigned int fixes compiler warnings.
........
................
r209056 | kpfleming | 2009-07-27 17:38:59 +0200 (Mon, 27 Jul 2009) | 10 lines
Restore explicit export of ASTCFLAGS/ASTLDFLAGS and underscore-variants to sub-makes.
During the recent Makefile improvements I made, it seemed the 'make' was
automatically carrying down the ASTCFLAGS/ASTLDFLAGS settings to sub-makes,
so I removed the explict export of them. However, there are some circumstances
where make does this, and some where it does not, so I've brought them back
to ensure they are always exported. I also removed an extraneous double setting
of _ASTLDFLAGS on *BSD platforms.
................
r209098 | dbrooks | 2009-07-27 18:33:50 +0200 (Mon, 27 Jul 2009) | 6 lines
Fixing typos. Replaces "recieved" with "received" and "initilize" with "initialize"
(closes issue #15571)
Reported by: alecdavis
................
r209132 | mmichelson | 2009-07-27 19:50:04 +0200 (Mon, 27 Jul 2009) | 24 lines
Merged revisions 209131 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r209131 | mmichelson | 2009-07-27 12:44:06 -0500 (Mon, 27 Jul 2009) | 18 lines
Allow for UDPTL to use only even-numbered ports if desired.
There are some VoIP providers out there that will not accept SDP
offers with odd numbered UDPTL ports. While it is my personal opinion
that these VoIP providers are misinterpreting RFC 2327, it really is
not a big deal to play along with their silly little games. Of course,
since restricting UDPTL ports to only even numbers reduces the range
of available ports by half, so the option to use only even port numbers
is off by default. A user can enable the behavior by setting
use_even_ports=yes in udptl.conf.
(closes issue #15182)
Reported by: CGMChris
Patches:
15182.patch uploaded by mmichelson (license 60)
Tested by: CGMChris
........
................
r209197 | mmichelson | 2009-07-27 22:11:42 +0200 (Mon, 27 Jul 2009) | 9 lines
Honor channel's music class when using realtime music on hold.
(closes issue #15051)
Reported by: alexh
Patches:
15051.patch uploaded by mmichelson (license 60)
Tested by: alexh
................
r209235 | mmichelson | 2009-07-27 22:54:54 +0200 (Mon, 27 Jul 2009) | 5 lines
Gracefully handle malformed RTP text packets.
AST-2009-004
................
r209256 | kpfleming | 2009-07-27 23:21:43 +0200 (Mon, 27 Jul 2009) | 10 lines
Make T.38 switchover in ReceiveFAX synchronous.
In receive mode, if the channel that ReceiveFAX is running on supports T.38,
we should *always* attempt to switch T.38, rather than listening for an incoming
CNG tone and only triggering on that. The channel may be using a low-bitrate
codec that distorts the CNG tone, the sending FAX endpoint may not send CNG
at all, or there could be a variety of other reasons that we don't detect it,
but in all those cases if T.38 is available we certainly want to use it.
................
r209279 | kpfleming | 2009-07-27 23:43:36 +0200 (Mon, 27 Jul 2009) | 7 lines
Cleanup T.38 negotiation changes.
Convert LOG_NOTICE messages about T.38 negotiation in debug level 1 messages,
clean up some looping logic, and correct an improper use of ast_free() for
freeing an ast_frame.
................
r209317 | tilghman | 2009-07-28 02:14:12 +0200 (Tue, 28 Jul 2009) | 9 lines
Merged revisions 209315 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r209315 | tilghman | 2009-07-27 19:12:03 -0500 (Mon, 27 Jul 2009) | 2 lines
Publish French extra sounds
........
................
r209331 | tilghman | 2009-07-28 02:20:26 +0200 (Tue, 28 Jul 2009) | 2 lines
Regex FTL
................
Modified:
team/phsultan/rtmp-support/ (props changed)
team/phsultan/rtmp-support/CHANGES
team/phsultan/rtmp-support/Makefile
team/phsultan/rtmp-support/Makefile.moddir_rules
team/phsultan/rtmp-support/Makefile.rules
team/phsultan/rtmp-support/UPGRADE.txt
team/phsultan/rtmp-support/addons/Makefile
team/phsultan/rtmp-support/addons/chan_mobile.c
team/phsultan/rtmp-support/agi/Makefile
team/phsultan/rtmp-support/apps/app_dial.c
team/phsultan/rtmp-support/apps/app_directed_pickup.c
team/phsultan/rtmp-support/apps/app_fax.c
team/phsultan/rtmp-support/apps/app_festival.c
team/phsultan/rtmp-support/apps/app_rpt.c
team/phsultan/rtmp-support/apps/app_voicemail.c
team/phsultan/rtmp-support/channels/Makefile
team/phsultan/rtmp-support/channels/chan_dahdi.c
team/phsultan/rtmp-support/channels/chan_iax2.c
team/phsultan/rtmp-support/channels/chan_misdn.c
team/phsultan/rtmp-support/channels/chan_sip.c
team/phsultan/rtmp-support/channels/chan_skinny.c
team/phsultan/rtmp-support/channels/chan_vpb.cc
team/phsultan/rtmp-support/channels/sig_analog.c
team/phsultan/rtmp-support/channels/sig_pri.c
team/phsultan/rtmp-support/channels/sig_pri.h
team/phsultan/rtmp-support/codecs/Makefile
team/phsultan/rtmp-support/codecs/gsm/Makefile
team/phsultan/rtmp-support/codecs/lpc10/Makefile
team/phsultan/rtmp-support/configs/chan_dahdi.conf.sample
team/phsultan/rtmp-support/configs/cli_aliases.conf.sample
team/phsultan/rtmp-support/configs/iax.conf.sample
team/phsultan/rtmp-support/configs/udptl.conf.sample
team/phsultan/rtmp-support/contrib/scripts/install_prereq
team/phsultan/rtmp-support/doc/CODING-GUIDELINES
team/phsultan/rtmp-support/doc/tex/odbcstorage.tex
team/phsultan/rtmp-support/doc/video_console.txt
team/phsultan/rtmp-support/doc/voicemail_odbc_postgresql.txt
team/phsultan/rtmp-support/funcs/Makefile
team/phsultan/rtmp-support/funcs/func_realtime.c
team/phsultan/rtmp-support/funcs/func_strings.c
team/phsultan/rtmp-support/funcs/func_timeout.c
team/phsultan/rtmp-support/include/asterisk/channel.h
team/phsultan/rtmp-support/include/asterisk/compat.h
team/phsultan/rtmp-support/include/asterisk/frame.h
team/phsultan/rtmp-support/include/asterisk/module.h
team/phsultan/rtmp-support/include/asterisk/udptl.h
team/phsultan/rtmp-support/main/Makefile
team/phsultan/rtmp-support/main/asterisk.exports
team/phsultan/rtmp-support/main/callerid.c
team/phsultan/rtmp-support/main/cel.c
team/phsultan/rtmp-support/main/channel.c
team/phsultan/rtmp-support/main/cli.c
team/phsultan/rtmp-support/main/db1-ast/Makefile
team/phsultan/rtmp-support/main/features.c
team/phsultan/rtmp-support/main/frame.c
team/phsultan/rtmp-support/main/loader.c
team/phsultan/rtmp-support/main/manager.c
team/phsultan/rtmp-support/main/rtp_engine.c
team/phsultan/rtmp-support/main/strcompat.c
team/phsultan/rtmp-support/main/translate.c
team/phsultan/rtmp-support/main/udptl.c
team/phsultan/rtmp-support/pbx/Makefile
team/phsultan/rtmp-support/pbx/pbx_dundi.c
team/phsultan/rtmp-support/res/Makefile
team/phsultan/rtmp-support/res/res_jabber.c
team/phsultan/rtmp-support/res/res_musiconhold.c
team/phsultan/rtmp-support/res/res_realtime.c
team/phsultan/rtmp-support/res/res_rtp_asterisk.c
team/phsultan/rtmp-support/res/res_smdi.c
team/phsultan/rtmp-support/sounds/sounds.xml
team/phsultan/rtmp-support/utils/Makefile
Propchange: team/phsultan/rtmp-support/
------------------------------------------------------------------------------
Binary property 'branch-1.4-blocked' - no diff available.
Propchange: team/phsultan/rtmp-support/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.
Propchange: team/phsultan/rtmp-support/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Tue Jul 28 07:36:37 2009
@@ -1,1 +1,1 @@
-/trunk:1-206803
+/trunk:1-209397
Modified: team/phsultan/rtmp-support/CHANGES
URL: http://svn.asterisk.org/svn-view/asterisk/team/phsultan/rtmp-support/CHANGES?view=diff&rev=209398&r1=209397&r2=209398
==============================================================================
--- team/phsultan/rtmp-support/CHANGES (original)
+++ team/phsultan/rtmp-support/CHANGES Tue Jul 28 07:36:37 2009
@@ -120,7 +120,8 @@
used by the rest of the system.
* Made use the nationalprefix and internationalprefix misdn.conf
parameters to prefix any received number from the ISDN link if that
- number has the corresponding Type-Of-Number.
+ number has the corresponding Type-Of-Number. NOTE: This includes
+ comparing the incoming call's dialed number against the MSN list.
* Added the following new parameters: unknownprefix, netspecificprefix,
subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
received number from the ISDN link if that number has the corresponding
Modified: team/phsultan/rtmp-support/Makefile
URL: http://svn.asterisk.org/svn-view/asterisk/team/phsultan/rtmp-support/Makefile?view=diff&rev=209398&r1=209397&r2=209398
==============================================================================
--- team/phsultan/rtmp-support/Makefile (original)
+++ team/phsultan/rtmp-support/Makefile Tue Jul 28 07:36:37 2009
@@ -13,29 +13,23 @@
# All Makefiles use the following variables:
#
-# ASTCFLAGS - compiler options
-# ASTLDFLAGS - linker flags (not libraries)
+# ASTCFLAGS - compiler options provided by the user (if any)
+# _ASTCFLAGS - compiler options provided by the build system
+# ASTLDFLAGS - linker flags (not libraries) provided by the user (if any)
[... 6005 lines stripped ...]
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