[asterisk-commits] eliel: branch eliel/data_api_providers_gsoc2010 r283593 - in /team/eliel/data...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Aug 25 13:25:56 CDT 2010
Author: eliel
Date: Wed Aug 25 13:25:45 2010
New Revision: 283593
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=283593
Log:
Merged revisions 282048,282067,282099,282132,282202,282237,282270,282272,282304,282335,282367,282397,282471,282502,282544,282546,282578,282609,282640,282673,282701,282751,282798,282827,282861,282892,282896,282980-282981,283015,283051,283174,283176,283178,283208,283210,283232,283242,283287-283289,283320,283351,283383,283456,283458,283494-283495,283528,283560,283562 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r282048 | dvossel | 2010-08-12 16:17:17 -0400 (Thu, 12 Aug 2010) | 42 lines
Merged revisions 282047 via svnmerge from
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r282047 | dvossel | 2010-08-12 15:15:41 -0500 (Thu, 12 Aug 2010) | 35 lines
improved translation paths for wideband codecs
The problem I'm addressing is that Asterisk's current
method of building the least cost translation paths
between codecs does not take into account sample rate.
For instance, it was possible for siren14 (a 32khz codec),
to contain the a translation path to siren7 (a 16khz
audio codec) that goes through slin at 8khz. In this
case Asterisk takes a 32khz codec, down samples it to
8khz and then up samples it to 16khz which is terrible
regardless if it is computationally less expensive. This
patch now builds translation paths that give priority to
maintaining the best possible sample rate before taking
into consideration computational cost. This patch also
adds cli commands to expose what translation paths are
actually being used.
Changes:
1. Translation paths will never contain a step that changes
the sample rate unless absolutely necessary.
2. When choosing the best codec to make two channels compatible.
Shared codecs with the highest sample rate are given priority.
3. A new cli command to show all translation paths available
for a specific codec 'core show translation paths [codec name]'
has been added.
4. 'core show translation' which displays the translation
matrix now includes the new higher bit audio codecs in the table.
5. 'core show channel [channel name]' now displays the
translation paths if translation is used.
(closes issue #16841)
Reported by: dvossel
Review: https://reviewboard.asterisk.org/r/842/
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r282067 | russell | 2010-08-12 16:44:39 -0400 (Thu, 12 Aug 2010) | 11 lines
Merged revisions 282066 via svnmerge from
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r282066 | russell | 2010-08-12 15:41:17 -0500 (Thu, 12 Aug 2010) | 4 lines
Add a "core reload" CLI command.
Review: https://reviewboard.asterisk.org/r/859/
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r282099 | rmudgett | 2010-08-12 18:10:49 -0400 (Thu, 12 Aug 2010) | 14 lines
Merged revisions 282098 via svnmerge from
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r282098 | rmudgett | 2010-08-12 17:06:06 -0500 (Thu, 12 Aug 2010) | 7 lines
Separate call completion config parameter allocation and default initialization.
If you ever have a need to reset the call completion config parameters
to defaults, now you can.
And no Virginia, C++ idioms do not always work in C.
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r282132 | qwell | 2010-08-12 18:52:33 -0400 (Thu, 12 Aug 2010) | 23 lines
Merged revisions 282131 via svnmerge from
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r282131 | qwell | 2010-08-12 17:51:44 -0500 (Thu, 12 Aug 2010) | 16 lines
Merged revisions 282130 via svnmerge from
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r282130 | qwell | 2010-08-12 17:50:54 -0500 (Thu, 12 Aug 2010) | 9 lines
Merged revisions 282129 via svnmerge from
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r282129 | qwell | 2010-08-12 17:49:28 -0500 (Thu, 12 Aug 2010) | 1 line
Register CLI commands before parsing config, in case there is a config error.
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r282202 | twilson | 2010-08-13 12:08:05 -0400 (Fri, 13 Aug 2010) | 21 lines
Merged revisions 282200-282201 via svnmerge from
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r282200 | twilson | 2010-08-13 11:00:02 -0500 (Fri, 13 Aug 2010) | 10 lines
Detect when libsrtp cannot be linked in a shared library
The libsrtp build system currently does not produce a shared library
or a static library compiled with -fPIC, so on 64-bit systems it is
possible that we will get a compile error if libsrtp is installed and
res_srtp is selected in menuselect.
This patch attempts to detect this situation and provide the user with
instructions to work around the problem.
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r282201 | twilson | 2010-08-13 11:02:20 -0500 (Fri, 13 Aug 2010) | 2 lines
Whitespace fix :-/
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r282237 | dvossel | 2010-08-13 14:58:49 -0400 (Fri, 13 Aug 2010) | 30 lines
Merged revisions 282236 via svnmerge from
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r282236 | dvossel | 2010-08-13 13:58:10 -0500 (Fri, 13 Aug 2010) | 23 lines
Merged revisions 282235 via svnmerge from
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r282235 | dvossel | 2010-08-13 13:54:53 -0500 (Fri, 13 Aug 2010) | 16 lines
only do magic pickup when notifycid is enabled
A new way of doing BLF pickup was introduced into 1.6.2. This feature
adds a call-id value into the XML of a SIP_NOTIFY message sent to alert
a subscriber that a device is ringing. This option should only be enabled
when the new 'notifycid' option is set... but this was not the case. Instead
the call-id value was included for every RINGING Notify message, which
caused a regression for people who used other methods for call pickup.
(closes issue #17633)
Reported by: urosh
Patches:
chan_sip.txt uploaded by urosh (license )
blf_cid_issue.diff uploaded by dvossel (license 671)
Tested by: dvossel, urosh, okrief, alecdavis
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r282270 | dvossel | 2010-08-13 16:05:44 -0400 (Fri, 13 Aug 2010) | 11 lines
Merged revisions 282269 via svnmerge from
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r282269 | dvossel | 2010-08-13 15:03:56 -0500 (Fri, 13 Aug 2010) | 4 lines
res_stun_monitor for monitoring network changes behind a NAT device
Review: https://reviewboard.asterisk.org/r/854
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r282272 | dvossel | 2010-08-13 16:12:22 -0400 (Fri, 13 Aug 2010) | 9 lines
Merged revisions 282271 via svnmerge from
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r282271 | dvossel | 2010-08-13 15:11:58 -0500 (Fri, 13 Aug 2010) | 2 lines
res_stun_monitor and corresponding options CHANGES documentation
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r282304 | dvossel | 2010-08-13 18:27:20 -0400 (Fri, 13 Aug 2010) | 17 lines
Merged revisions 282302 via svnmerge from
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r282302 | dvossel | 2010-08-13 17:23:38 -0500 (Fri, 13 Aug 2010) | 10 lines
remove current STUN support from chan_sip.c
This patch removes the current broken/useless stun
support from chan_sip.
(closes issue #17622)
Reported by: philipp2
Review: https://reviewboard.asterisk.org/r/855/
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r282335 | rmudgett | 2010-08-13 19:57:21 -0400 (Fri, 13 Aug 2010) | 13 lines
Merged revisions 282334 via svnmerge from
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r282334 | rmudgett | 2010-08-13 18:53:36 -0500 (Fri, 13 Aug 2010) | 6 lines
PRI CCSS may use a stale dial string for the recall dial string.
If an outgoing call negotiates a different B channel than initially
requested, the saved original dial string was not transferred to the new B
channel. CCSS uses that dial string to generate the recall dial string.
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r282367 | tilghman | 2010-08-14 00:58:34 -0400 (Sat, 14 Aug 2010) | 11 lines
Merged revisions 282366 via svnmerge from
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r282366 | tilghman | 2010-08-13 23:53:58 -0500 (Fri, 13 Aug 2010) | 4 lines
Fix our FRACKing issue with chan_iax2 a different way.
Review: https://reviewboard.asterisk.org/r/861/
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r282397 | tzafrir | 2010-08-15 09:08:45 -0400 (Sun, 15 Aug 2010) | 16 lines
Support for GNU/kFreeBSD
kFreeBSD is GNU (with glibc) on to of a FreeBSD kernel. See
http://glibc-bsd.alioth.debian.org/porting/PORTING
This patch gets Asterisk close to building on Debian kFreeBSD i386,
mainly by adding an extra test for __GLIBC__ in one or two (or more)
places.
OSARCH is set to 'kfreebsd-gnu'
DAHDI support (and support for chan_vpb) was not tested.
Review: https://reviewboard.asterisk.org/r/858/
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r282471 | lmadsen | 2010-08-16 14:02:29 -0400 (Mon, 16 Aug 2010) | 22 lines
Merged revisions 282470 via svnmerge from
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r282470 | lmadsen | 2010-08-16 13:01:00 -0500 (Mon, 16 Aug 2010) | 15 lines
Merged revisions 282469 via svnmerge from
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r282469 | lmadsen | 2010-08-16 13:00:09 -0500 (Mon, 16 Aug 2010) | 7 lines
Add information about creating sounds files using
the sounds tools publically available so that others can create their
own sounds prompts using the same tools we use to generate sounds releases.
This allows people creating their own prompts to sound consistent with
the prompts available from the open source project.
SWP-595
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r282502 | twilson | 2010-08-16 16:40:55 -0400 (Mon, 16 Aug 2010) | 37 lines
Merged revisions 282468 via svnmerge from
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r282468 | twilson | 2010-08-16 12:53:44 -0500 (Mon, 16 Aug 2010) | 30 lines
Merged revisions 282467 via svnmerge from
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r282467 | twilson | 2010-08-16 12:32:01 -0500 (Mon, 16 Aug 2010) | 23 lines
Merged revisions 282430 via svnmerge from
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r282430 | twilson | 2010-08-16 12:06:37 -0500 (Mon, 16 Aug 2010) | 16 lines
Send a SRCCHANGE indication when we masquerade
Masquerading a channel means that the src of the audio is potentially
changing, so send a SRCCHANGE so that RTP-based media streams can get
a new SSRC generated to reflect the change. Original patch by addix
(along with lots of testing--thanks!).
(closes issue #17007)
Reported by: addix
Patches:
1001-reset-SSRC-original-channel.diff uploaded by addix (license 1006)
srcchange.diff uploaded by twilson (license 396)
Tested by: addix, twilson
Review: https://reviewboard.asterisk.org/r/862/
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r282544 | dvossel | 2010-08-17 15:34:52 -0400 (Tue, 17 Aug 2010) | 11 lines
Merged revisions 282543 via svnmerge from
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r282543 | dvossel | 2010-08-17 14:34:06 -0500 (Tue, 17 Aug 2010) | 4 lines
fixes truncated uint64_t value in put_unaligned_uint64_t() function
(issue #17804)
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r282546 | dvossel | 2010-08-17 16:09:30 -0400 (Tue, 17 Aug 2010) | 12 lines
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r282545 | dvossel | 2010-08-17 15:08:56 -0500 (Tue, 17 Aug 2010) | 6 lines
ACCEPT message should respond with the new FORMAT2 ie
(closes issue #17804)
Reported by: tpanton
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r282578 | dvossel | 2010-08-17 17:37:46 -0400 (Tue, 17 Aug 2010) | 23 lines
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r282577 | dvossel | 2010-08-17 16:36:57 -0500 (Tue, 17 Aug 2010) | 16 lines
Merged revisions 282576 via svnmerge from
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r282576 | dvossel | 2010-08-17 16:35:17 -0500 (Tue, 17 Aug 2010) | 9 lines
fixes no default transport for temp peer creation in chan_sip
(closes issue #17829)
Reported by: falves11
Patches:
issue_17829.rev1.txt uploaded by russell (license 2)
issue_17829.diff uploaded by dvossel (license 671)
Tested by: falves11
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r282609 | tilghman | 2010-08-18 03:50:07 -0400 (Wed, 18 Aug 2010) | 23 lines
Merged revisions 282608 via svnmerge from
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r282608 | tilghman | 2010-08-18 02:49:04 -0500 (Wed, 18 Aug 2010) | 16 lines
Merged revisions 282607 via svnmerge from
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r282607 | tilghman | 2010-08-18 02:43:14 -0500 (Wed, 18 Aug 2010) | 9 lines
Don't warn on callerid when completely text, instead of numeric with localdialplan prefixes.
(closes issue #16770)
Reported by: jamicque
Patches:
20100413__issue16770.diff.txt uploaded by tilghman (license 14)
20100811__issue16770.diff.txt uploaded by tilghman (license 14)
Tested by: jamicque
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r282640 | mnicholson | 2010-08-18 09:11:38 -0400 (Wed, 18 Aug 2010) | 20 lines
Merged revisions 282639 via svnmerge from
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r282639 | mnicholson | 2010-08-18 08:10:39 -0500 (Wed, 18 Aug 2010) | 13 lines
Properly handle 200 and unknown responses conatined in NOTIFY requests received in response to REFER requests.
This patch fixes the way asterisk handles NOTIFY requests received in response to REFER requests. These changes to NOTIFY handler were first introduced in r217482. This new change properly handles the 200 response by queueing an AST_TRANSFER_SUCCESS control frame and also prevents that control frame from being queued when provisional and unknown responses are received.
(issue #17486)
Reported by: davidw
Tested by: mnicholson
(issue #12713)
Reported by: davidw
Review: https://reviewboard.asterisk.org/r/860/
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r282673 | rmudgett | 2010-08-18 11:35:23 -0400 (Wed, 18 Aug 2010) | 13 lines
Merged revisions 282671-282672 via svnmerge from
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r282671 | rmudgett | 2010-08-18 10:27:51 -0500 (Wed, 18 Aug 2010) | 1 line
Use the correct operator when calculating the PRI span devstate.
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r282672 | rmudgett | 2010-08-18 10:28:27 -0500 (Wed, 18 Aug 2010) | 1 line
Use the correct type for aoce_delayhangup bit field.
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r282701 | wedhorn | 2010-08-18 17:34:27 -0400 (Wed, 18 Aug 2010) | 12 lines
Cleanup: consolidate offhook (new call).
Consolidates all offhook (new call with dialtone) to setsubstate_offhook. This should be roughly equivalent to existing code, although a couple of calls now run through the full offhook sequence rather than an abbreviated one.
(closes issue #17812)
Reported by: wedhorn
Patches:
cleanup.stateoffhook.diff uploaded by wedhorn (license 30)
Tested by: salecha, wedhorn
Review: NA
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r282751 | twilson | 2010-08-18 22:20:42 -0400 (Wed, 18 Aug 2010) | 23 lines
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r282740 | twilson | 2010-08-18 21:18:50 -0500 (Wed, 18 Aug 2010) | 16 lines
Merged revisions 282730 via svnmerge from
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r282730 | twilson | 2010-08-18 21:14:28 -0500 (Wed, 18 Aug 2010) | 9 lines
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r282729 | twilson | 2010-08-18 21:12:55 -0500 (Wed, 18 Aug 2010) | 2 lines
Add some documentation about codec negotiation to sip.conf
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r282798 | russell | 2010-08-19 08:13:41 -0400 (Thu, 19 Aug 2010) | 2 lines
Add a todo item for CEL.
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r282827 | tilghman | 2010-08-19 10:46:08 -0400 (Thu, 19 Aug 2010) | 9 lines
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r282826 | tilghman | 2010-08-19 09:44:51 -0500 (Thu, 19 Aug 2010) | 2 lines
Only output debugging if the debug level is on.
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r282861 | mnicholson | 2010-08-19 16:02:52 -0400 (Thu, 19 Aug 2010) | 37 lines
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r282860 | mnicholson | 2010-08-19 15:01:11 -0500 (Thu, 19 Aug 2010) | 30 lines
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r282859 | mnicholson | 2010-08-19 14:44:00 -0500 (Thu, 19 Aug 2010) | 23 lines
Merged revisions 277944 via svnmerge from
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r277944 | pabelanger | 2010-07-19 15:56:07 -0500 (Mon, 19 Jul 2010) | 16 lines
Regression with T.38 negotiation
Prior to 1.4.26.3 T.38 negotiation worked properly, in the case
of the reporter.
(issue #16852)
Reported by: cfc
(closes issue #16705)
Reported by: mpiazzatnetbug
Patches:
issue16705_2.diff uploaded by ebroad (license 878)
Tested by: vrban, ebroad, c0rnoTa, samdell3
Review: https://reviewboard.asterisk.org/r/754/
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r282892 | dvossel | 2010-08-19 16:35:42 -0400 (Thu, 19 Aug 2010) | 18 lines
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r282891 | dvossel | 2010-08-19 15:34:41 -0500 (Thu, 19 Aug 2010) | 11 lines
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r282890 | dvossel | 2010-08-19 15:31:22 -0500 (Thu, 19 Aug 2010) | 5 lines
fixes sip peer memory leaks in the peer_by_ip table
(issue #17798)
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r282896 | dvossel | 2010-08-19 17:08:39 -0400 (Thu, 19 Aug 2010) | 32 lines
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r282895 | dvossel | 2010-08-19 16:07:20 -0500 (Thu, 19 Aug 2010) | 25 lines
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r282894 | dvossel | 2010-08-19 16:05:54 -0500 (Thu, 19 Aug 2010) | 18 lines
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r282893 | dvossel | 2010-08-19 16:03:24 -0500 (Thu, 19 Aug 2010) | 11 lines
tos_sip option was not being set correctly
When tos_sip is used, the tos of the sip socket is only set
correctly if the socket binding changes on a reload. If the binding
stays the same but the TOS changes, the new tos value would not take
into effect. This patch fixes that.
(closes issue #17712)
Reported by: nickb
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r282980 | russell | 2010-08-20 07:54:00 -0400 (Fri, 20 Aug 2010) | 11 lines
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r282638 | russell | 2010-08-18 07:30:40 -0500 (Wed, 18 Aug 2010) | 4 lines
Split _all_ arguments before parsing them.
This fixes multicast RTP paging using linksys mode.
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r282981 | russell | 2010-08-20 07:54:22 -0400 (Fri, 20 Aug 2010) | 9 lines
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r282979 | russell | 2010-08-20 06:52:37 -0500 (Fri, 20 Aug 2010) | 2 lines
Add an argument missing from the CELGenUserEvent documentation.
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r283015 | russell | 2010-08-20 08:45:30 -0400 (Fri, 20 Aug 2010) | 9 lines
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r283013 | russell | 2010-08-20 07:45:12 -0500 (Fri, 20 Aug 2010) | 2 lines
Fix a typo in a column name.
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r283051 | rmudgett | 2010-08-20 11:39:25 -0400 (Fri, 20 Aug 2010) | 43 lines
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r283050 | rmudgett | 2010-08-20 10:35:38 -0500 (Fri, 20 Aug 2010) | 36 lines
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r283049 | rmudgett | 2010-08-20 10:31:03 -0500 (Fri, 20 Aug 2010) | 29 lines
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r283048 | rmudgett | 2010-08-20 10:24:36 -0500 (Fri, 20 Aug 2010) | 22 lines
Q931 - Sending PROGRESS after sending ALERTING is a protocol error
The PRI layer in chan_dadhi will check if a PROGRESS message has already
been sent, and not allow sending another (although that is technically
allowed by the Q931 spec), however it does not protect against sending an
ALERTING and then sending a PROGRESS message, which is a violation of the
specification.
Most switches don't seem to care too deeply about this, but some do, and
will disconnect the call when receiving this invalid sequence.
Protocol specification reference: T-REC-Q.931-199805-I page 223, "Figure
A.5/Q.931 -- Overview protocol control (network side) point-point
(sheet 3 of 8)"
(closes issue #17874)
Reported by: nic_bellamy
Patches:
asterisk-1.4-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299)
asterisk-1.6.2-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299)
asterisk-trunk-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299)
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r283174 | russell | 2010-08-23 07:59:51 -0400 (Mon, 23 Aug 2010) | 12 lines
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r283173 | russell | 2010-08-23 06:58:34 -0500 (Mon, 23 Aug 2010) | 5 lines
Expand cel_custom.conf.sample.
Include the usage of CSV_QUOTE() to ensure data has valid CSV formatting. Also list
the special CEL variables that are available for use in the mapping.
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r283176 | tilghman | 2010-08-23 08:09:17 -0400 (Mon, 23 Aug 2010) | 9 lines
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r283175 | tilghman | 2010-08-23 07:06:26 -0500 (Mon, 23 Aug 2010) | 2 lines
Don't fail to start if the config file is missing.
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r283178 | russell | 2010-08-23 08:13:16 -0400 (Mon, 23 Aug 2010) | 9 lines
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r283177 | russell | 2010-08-23 07:12:53 -0500 (Mon, 23 Aug 2010) | 2 lines
Cut down on excessive quotation.
........
................
r283208 | russell | 2010-08-23 08:33:24 -0400 (Mon, 23 Aug 2010) | 9 lines
Merged revisions 283207 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r283207 | russell | 2010-08-23 07:31:20 -0500 (Mon, 23 Aug 2010) | 2 lines
Tack on ${eventextra} to the sample cel_custom.conf.
........
................
r283210 | russell | 2010-08-23 09:09:47 -0400 (Mon, 23 Aug 2010) | 9 lines
Merged revisions 283209 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r283209 | russell | 2010-08-23 08:06:57 -0500 (Mon, 23 Aug 2010) | 2 lines
Don't blow up on an invalid AMA flag.
........
................
r283232 | russell | 2010-08-23 09:23:37 -0400 (Mon, 23 Aug 2010) | 14 lines
Merged revisions 283230 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r283230 | russell | 2010-08-23 08:23:12 -0500 (Mon, 23 Aug 2010) | 7 lines
Make the AST_CEL_AMA enum match up with the AST_CDR_ ama flag values.
Really, having 2 enums for this is silly and error prone, demonstrated by
the crash that I hit because there was an assumption in the code that the
values in each matched up. However, this is a quick fix to get them to
match up so it will work.
........
................
r283242 | russell | 2010-08-23 09:35:55 -0400 (Mon, 23 Aug 2010) | 9 lines
Merged revisions 283241 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r283241 | russell | 2010-08-23 08:35:35 -0500 (Mon, 23 Aug 2010) | 2 lines
Add sample configuration for cel_radius.
........
................
r283287 | wedhorn | 2010-08-23 16:23:51 -0400 (Mon, 23 Aug 2010) | 12 lines
Fixes display issues on 7910 and older phones.
Also correct the callinfo provided in skinny_answer.
(closes issue #17876)
Reported by: salecha
Patches:
skinny_cnd3.diff uploaded by wedhorn (license 30)
Tested by: salecha, wedhorn
Review: NA
................
r283288 | wedhorn | 2010-08-23 16:39:05 -0400 (Mon, 23 Aug 2010) | 2 lines
Add additional AST_CONTROL_ states to control2str.
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r283289 | wedhorn | 2010-08-23 16:50:55 -0400 (Mon, 23 Aug 2010) | 2 lines
Hack to allow easy debugging of skinny in trunk.
................
r283320 | tilghman | 2010-08-23 17:35:09 -0400 (Mon, 23 Aug 2010) | 16 lines
Merged revisions 283319 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r283319 | tilghman | 2010-08-23 16:33:47 -0500 (Mon, 23 Aug 2010) | 9 lines
Merged revisions 283318 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r283318 | tilghman | 2010-08-23 16:32:14 -0500 (Mon, 23 Aug 2010) | 2 lines
CDR drivers depend upon res_odbc, not directly on the ODBC libraries
........
................
................
r283351 | russell | 2010-08-24 08:51:46 -0400 (Tue, 24 Aug 2010) | 9 lines
Merged revisions 283350 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r283350 | russell | 2010-08-24 07:49:41 -0500 (Tue, 24 Aug 2010) | 2 lines
Don't attempt to release a NULL ODBC handle.
........
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r283383 | dvossel | 2010-08-24 12:12:36 -0400 (Tue, 24 Aug 2010) | 32 lines
Merged revisions 283382 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r283382 | dvossel | 2010-08-24 11:11:18 -0500 (Tue, 24 Aug 2010) | 25 lines
Merged revisions 283381 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r283381 | dvossel | 2010-08-24 11:07:37 -0500 (Tue, 24 Aug 2010) | 18 lines
Merged revisions 283380 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r283380 | dvossel | 2010-08-24 11:01:51 -0500 (Tue, 24 Aug 2010) | 11 lines
This fix makes sure the ast_channel hangs up correctly when the dialog's PENDING_BYE flag is set.
When the pending bye flag is used, it is possible that the dialog will terminate
and leave the sip_pvt->owner channel up. This is because we never hangup the
ast_channel after sending the SIP_BYE request. When we receive the response for
the SIP_BYE we set need_destroy which we would expect to destroy the dialog on the
next do_monitor loop, but this is not the case. The dialog will only be destroyed
once the owner is hungup even with the need_destroy flag set. This patch sets the
softhangup flag on the ast_channel when a SIP_BYE request is sent as a result of the
pending bye flag.
........
................
................
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r283456 | dvossel | 2010-08-24 14:45:57 -0400 (Tue, 24 Aug 2010) | 2 lines
This fix downgrades the ERROR message indicating no res_stun_monitor.conf to a WARNING message.
................
r283458 | lmadsen | 2010-08-24 14:58:46 -0400 (Tue, 24 Aug 2010) | 17 lines
Merged revisions 283457 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r283457 | lmadsen | 2010-08-24 13:56:29 -0500 (Tue, 24 Aug 2010) | 9 lines
Fix issue where TOS is no longer set on RTP packets.
Fix issue where the tos is no longer being set on RTP packets through res_rtp_asterisk.
(closes issue #17890)
Reported by: elguero
Patches:
qos_18.diff uploaded by elguero (license 37)
Review: https://reviewboard.asterisk.org/r/868
........
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r283494 | dvossel | 2010-08-24 16:36:35 -0400 (Tue, 24 Aug 2010) | 9 lines
Merged revisions 283493 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r283493 | dvossel | 2010-08-24 15:34:03 -0500 (Tue, 24 Aug 2010) | 2 lines
Changes the default behavior for sip.conf's pedantic option from "no" to "yes".
........
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r283495 | wedhorn | 2010-08-24 16:42:30 -0400 (Tue, 24 Aug 2010) | 8 lines
Ignore redial hard button when no previous number.
(closes issue #17887)
Reported by: salecha
Patches:
skinny.redial.diff uploaded by wedhorn (license 30)
Tested by: wedhorn, salecha
................
r283528 | russell | 2010-08-25 10:55:47 -0400 (Wed, 25 Aug 2010) | 9 lines
Merged revisions 283527 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r283527 | russell | 2010-08-25 09:55:00 -0500 (Wed, 25 Aug 2010) | 2 lines
Convert ast_log(LOG_DEBUG, ...) to ast_debug(...)
........
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r283560 | dvossel | 2010-08-25 11:56:05 -0400 (Wed, 25 Aug 2010) | 23 lines
Merged revisions 283559 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r283559 | dvossel | 2010-08-25 10:54:11 -0500 (Wed, 25 Aug 2010) | 16 lines
Merged revisions 283558 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r283558 | dvossel | 2010-08-25 10:52:54 -0500 (Wed, 25 Aug 2010) | 10 lines
Asterisk will not advertise session timers are supported when 'session-timers=refuse' is used.
Asterisk now dynamically builds the "Supported" header depending
on what is enabled/disabled in sip.conf. Session timers used
to always be advertised as being supported even when they were disabled
in the configuration. This caused problems with some end points.
(issue #17005)
........
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r283562 | tilghman | 2010-08-25 12:14:11 -0400 (Wed, 25 Aug 2010) | 12 lines
Merged revisions 283561 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r283561 | tilghman | 2010-08-25 11:12:43 -0500 (Wed, 25 Aug 2010) | 5 lines
Initialize connect timeout on each time through the loop.
(closes issue #17911)
Reported by: wurstsalat
........
................
Modified:
team/eliel/data_api_providers_gsoc2010/ (props changed)
team/eliel/data_api_providers_gsoc2010/CHANGES
team/eliel/data_api_providers_gsoc2010/UPGRADE-1.8.txt
team/eliel/data_api_providers_gsoc2010/apps/app_celgenuserevent.c
team/eliel/data_api_providers_gsoc2010/cdr/cdr_adaptive_odbc.c
team/eliel/data_api_providers_gsoc2010/cdr/cdr_odbc.c
team/eliel/data_api_providers_gsoc2010/cel/cel_adaptive_odbc.c
team/eliel/data_api_providers_gsoc2010/channels/chan_dahdi.c
team/eliel/data_api_providers_gsoc2010/channels/chan_iax2.c
team/eliel/data_api_providers_gsoc2010/channels/chan_multicast_rtp.c
team/eliel/data_api_providers_gsoc2010/channels/chan_oss.c
team/eliel/data_api_providers_gsoc2010/channels/chan_sip.c
team/eliel/data_api_providers_gsoc2010/channels/chan_skinny.c
team/eliel/data_api_providers_gsoc2010/channels/sig_pri.c
team/eliel/data_api_providers_gsoc2010/channels/sig_pri.h
team/eliel/data_api_providers_gsoc2010/channels/sip/include/sip.h
team/eliel/data_api_providers_gsoc2010/configs/cel.conf.sample
team/eliel/data_api_providers_gsoc2010/configs/cel_adaptive_odbc.conf.sample
team/eliel/data_api_providers_gsoc2010/configs/cel_custom.conf.sample
team/eliel/data_api_providers_gsoc2010/configs/iax.conf.sample
team/eliel/data_api_providers_gsoc2010/configs/sip.conf.sample
team/eliel/data_api_providers_gsoc2010/configure
team/eliel/data_api_providers_gsoc2010/configure.ac
team/eliel/data_api_providers_gsoc2010/doc/tex/asterisk.tex
team/eliel/data_api_providers_gsoc2010/funcs/func_odbc.c
team/eliel/data_api_providers_gsoc2010/include/asterisk/ccss.h
team/eliel/data_api_providers_gsoc2010/include/asterisk/cel.h
team/eliel/data_api_providers_gsoc2010/include/asterisk/event_defs.h
team/eliel/data_api_providers_gsoc2010/include/asterisk/sched.h
team/eliel/data_api_providers_gsoc2010/include/asterisk/translate.h
team/eliel/data_api_providers_gsoc2010/include/asterisk/unaligned.h
team/eliel/data_api_providers_gsoc2010/main/Makefile
team/eliel/data_api_providers_gsoc2010/main/acl.c
team/eliel/data_api_providers_gsoc2010/main/ccss.c
team/eliel/data_api_providers_gsoc2010/main/cel.c
team/eliel/data_api_providers_gsoc2010/main/channel.c
team/eliel/data_api_providers_gsoc2010/main/cli.c
team/eliel/data_api_providers_gsoc2010/main/netsock.c
team/eliel/data_api_providers_gsoc2010/main/netsock2.c
team/eliel/data_api_providers_gsoc2010/main/translate.c
team/eliel/data_api_providers_gsoc2010/pbx/pbx_config.c
team/eliel/data_api_providers_gsoc2010/res/res_odbc.c
team/eliel/data_api_providers_gsoc2010/res/res_rtp_asterisk.c
team/eliel/data_api_providers_gsoc2010/utils/muted.c
Propchange: team/eliel/data_api_providers_gsoc2010/
------------------------------------------------------------------------------
Binary property 'branch-1.8-merged' - no diff available.
Propchange: team/eliel/data_api_providers_gsoc2010/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Wed Aug 25 13:25:45 2010
@@ -1,1 +1,1 @@
-/trunk:1-282018
+/trunk:1-283592
Modified: team/eliel/data_api_providers_gsoc2010/CHANGES
URL: http://svnview.digium.com/svn/asterisk/team/eliel/data_api_providers_gsoc2010/CHANGES?view=diff&rev=283593&r1=283592&r2=283593
==============================================================================
--- team/eliel/data_api_providers_gsoc2010/CHANGES (original)
+++ team/eliel/data_api_providers_gsoc2010/CHANGES Wed Aug 25 13:25:45 2010
@@ -81,6 +81,8 @@
RTP has been outfitted with the same abilities.
* Added support for setting the Max-Forwards: header in SIP requests. Setting is
available in device configurations as well as in the dial plan.
+ * Addition of the 'subscribe_network_change' option for turning on and off
+ res_stun_monitor module support in chan_sip.
IAX2 Changes
-----------
@@ -90,6 +92,9 @@
encryption is being used. This interoperates with the SIP SRTP implementation
so that a secure SIP call can be bridged to a secure IAX call when the
dialplan requires bridged channels to be "secure".
+ * Addition of the 'subscribe_network_change' option for turning on and off
+ res_stun_monitor module support in chan_iax.
+
MGCP Changes
------------
@@ -543,6 +548,10 @@
* The UNISTIM channel driver (chan_unistim) has been updated to support devices that
have less than 3 lines on the LCD.
* Realtime now supports database failover. See the sample extconfig.conf for details.
+ * The addition of improved translation path building for wideband codecs. Sample
+ rate changes during translation are now avoided unless absolutely necessary.
+ * The addition of the res_stun_monitor module for monitoring and reacting to network
+ changes while behind a NAT.
CLI Changes
-----------
@@ -558,6 +567,7 @@
manager.conf.
* Added 'all' keyword to the CLI command "channel request hangup" so that you can send
the channel hangup request to all channels.
+ * Added a "core reload" CLI command that executes a global reload of Asterisk.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
@@ -1099,10 +1109,6 @@
option is enabled, Asterisk will watch for a CNG tone in the incoming audio
for a received call. If it is detected, the channel will jump to the
'fax' extension in the dialplan.
- * Improved NAT and STUN support.
- chan_sip now can use port numbers in bindaddr, externip and externhost
- options, as well as contact a STUN server to detect its external address
- for the SIP socket. See sip.conf.sample, 'NAT' section.
* The default SIP useragent= identifier now includes the Asterisk version
* A new option, match_auth_username in sip.conf changes the matching of incoming requests.
If set, and the incoming request carries authentication info,
Modified: team/eliel/data_api_providers_gsoc2010/UPGRADE-1.8.txt
URL: http://svnview.digium.com/svn/asterisk/team/eliel/data_api_providers_gsoc2010/UPGRADE-1.8.txt?view=diff&rev=283593&r1=283592&r2=283593
[... 2775 lines stripped ...]
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