Age | Commit message (Collapse) | Author |
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Moving a found object to the front of its bucket to reduce the necessary traversal steps to find an object. This change improves the search time on large system with many data or in link lists.
(closes issue #19233)
Reported by: schmidts
Review: https://reviewboard.asterisk.org/r/1201/
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r316917 | seanbright | 2011-05-04 22:23:28 -0400 (Wed, 04 May 2011) | 5 lines
Make sure that tcptls_session is properly initialized.
(issue #18598)
Reported by: ksn
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r316918 | seanbright | 2011-05-04 22:25:20 -0400 (Wed, 04 May 2011) | 5 lines
Look at the correct buffer for our digest info instead of an empty one.
(issue #18598)
Reported by: ksn
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r316919 | seanbright | 2011-05-04 22:30:45 -0400 (Wed, 04 May 2011) | 10 lines
Use the correct HTTP method when generating our digest, otherwise we always fail.
When calculating the 'A2' portion of our digest for verification, we need the
HTTP method that is currently in use. Unfortunately our mapping function was
incorrect, resulting in invalid hashes being generated and, in turn, failures
in authentication.
(closes issue #18598)
Reported by: ksn
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r316663 | seanbright | 2011-05-04 10:35:05 -0400 (Wed, 04 May 2011) | 8 lines
Only return a single error via AMI when requesting a forbidden action.
(closes issue #19216)
Reported by: oej
Patches:
issue19216-1.8-r316204.patch uploaded by seanbright (license 71)
Tested by: seanbright
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r316334 | dvossel | 2011-05-03 17:05:59 -0500 (Tue, 03 May 2011) | 8 lines
Fixes framehook segfault on indicate
(closes issue #19215)
Reported by: irroot
Patches:
framehook_indicate.patch uploaded by irroot (license 52)
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r316265 | russell | 2011-05-03 14:55:49 -0500 (Tue, 03 May 2011) | 5 lines
Fix a bunch of compiler warnings generated by gcc 4.6.0.
Most of these are -Wunused-but-set-variable, but there were a few others
mixed in here, as well.
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r316206 | seanbright | 2011-05-03 14:17:36 -0400 (Tue, 03 May 2011) | 8 lines
If we aren't interested in events, don't generate the FullyBooted event on AMI login.
(closes issue #19089)
Reported by: bklang
Patches:
issue19089-1.8-r316204.patch uploaded by seanbright (license 71)
Tested by: seanbright
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r315810 | russell | 2011-04-27 10:55:48 -0500 (Wed, 27 Apr 2011) | 2 lines
Set the copyright year to 2011 in the startup message.
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r315644 | twilson | 2011-04-26 14:39:01 -0700 (Tue, 26 Apr 2011) | 32 lines
Merged revisions 315643 via svnmerge from
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r315643 | twilson | 2011-04-26 14:27:44 -0700 (Tue, 26 Apr 2011) | 25 lines
Merged revisions 315596 via svnmerge from
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r315596 | twilson | 2011-04-26 14:16:10 -0700 (Tue, 26 Apr 2011) | 18 lines
Allow transfer loops without allowing forwarding loops
We try to avoid the situation where two phones may be forwarded to each other
causing an infinite loop by storing each dialed interface in a channel
datastore and checking the list before dialing out. This works, but currently
breaks situations like A calls B, A transfers B to C, B transfers C to A, and A
transfers C to B. Since human interaction is happening here and not an
automated forwarding loop, it should be allowed.
This patch removes the dialed_interfaces datastore when a call is bridged (a
suggestion from the brilliant mmichelson). If a call is being bridged, it
should be safe to assume that we aren't stuck in a loop.
Since we are now handling this is the bridge code, the previous attempts at
handling it in app_dial and app_queue are removed.
Review: https://reviewboard.asterisk.org/r/1195/
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r315645 | rmudgett | 2011-04-26 17:14:31 -0500 (Tue, 26 Apr 2011) | 21 lines
The 'e' special extension fails to trigger in at least two cases.
The 'e' extension is a fall back for the 'i', 't', or 'T' extensions if
any of them do not exist. Many of the places the 'e' extension was
supposed to be invoked fail because the priority was set wrong. There
were two places where the 'e' extension was not even checked for fall
back.
* Made invoke the 'e' extension similarly to the previous 'i', 't', or 'T'
extension check and added the 'e' extension as a fall back to the two
missing locations.
* Prioritized and optimized some hangup tests associated with the 'e'
extension.
(closes issue #19136)
Reported by: kshumard
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1196/
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r314628 | mnicholson | 2011-04-21 13:24:05 -0500 (Thu, 21 Apr 2011) | 27 lines
Merged revisions 314620 via svnmerge from
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r314620 | mnicholson | 2011-04-21 13:22:19 -0500 (Thu, 21 Apr 2011) | 20 lines
Merged revisions 314607 via svnmerge from
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r314607 | mnicholson | 2011-04-21 13:19:21 -0500 (Thu, 21 Apr 2011) | 14 lines
Added limits to the number of unauthenticated sessions TCP based protocols are allowed to have open simultaneously. Also added timeouts for unauthenticated sessions where it made sense to do so.
Unrelated, the manager interface now properly checks if the user has the "system" privilege before executing shell commands via the Originate action.
AST-2011-005
AST-2011-006
(closes issue #18787)
Reported by: kobaz
(related to issue #18996)
Reported by: tzafrir
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Includes a new highly optimized and customizable
ConfBridge application capable of mixing audio at
sample rates ranging from 8khz-192khz.
Review: https://reviewboard.asterisk.org/r/1147/
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Review: https://reviewboard.asterisk.org/r/1157/
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r314358 | twilson | 2011-04-19 22:25:15 -0700 (Tue, 19 Apr 2011) | 4 lines
Initialize track pointer
ast_reentrancy_init checks to see if it is NULL before initializing with calloc
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r314251 | lmadsen | 2011-04-19 10:42:10 -0500 (Tue, 19 Apr 2011) | 8 lines
Use SSLv23_client_method instead of old SSLv2 only.
(closes issue #19095)
(closes issue #19138)
Reported by: tzafrir
Patches:
no_ssl2.diff uploaded by tzafrir (license 46)
Tested by: russell, chazzam
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r314017 | dvossel | 2011-04-18 08:41:06 -0500 (Mon, 18 Apr 2011) | 17 lines
sip codec negotiation of dynamic rtp payloads error fix
This patch fixes how chan_sip handles dynamic rtp payload types
it does not understand. At the moment if a dynamic payload's mime
type does not match one we understand, the payload does not get
removed from our payload table. As a result of this, the payload
is set to whatever dynamic codec we use internally for that payload
number on outgoing INVITES. This is incorrect.
This patch fixes this by properly checking the rtpmap set function's
return code to make sure it was found. The function can return both
-1 and -2 depending on the source of the mismatch. We were just
checking -1 explicitly.
Review: https://reviewboard.asterisk.org/r/1169/
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r313860 | jrose | 2011-04-15 10:08:05 -0500 (Fri, 15 Apr 2011) | 17 lines
Merged revisions 313859 via svnmerge from
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r313859 | jrose | 2011-04-15 09:58:37 -0500 (Fri, 15 Apr 2011) | 10 lines
Fix a Tab Completion bug that occurs due to multiple matches on a substring.
Makes word_match function in cli.c repeat a search for a command string until
a proper match is found or the string is searched to the last point.
(closes issue #17494)
Reported by: ffossard
Review: https://reviewboard.asterisk.org/r/1180/
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Add Asterisk Device State information and callbacks to the Call Completion
Supplemental Services for generic agents.
There are currently not many devices that have native support for CCSS.
Even as the devices become available there may be other reasons why one
may choose to not take advantage of the native abilities and stick with
the generic implementation. The generic implementation is quite capable
and could be greatly enhanced by adding device state capabilities. A
phone could then subscribe to the device state with a BLF key in
conjunction with Asterisk hints.
The advantages of the device state information would allow a single button
to: request CCSS, cancel a CCSS request, and display the current state of
a CCSS request.
For example, you may have a single button that when not lit, there is no
active CCSS request. When you press that button, the dialplan can query
the DEVICE_STATE() associated with that caller to determine whether they
should be calling CallCompletionRequest() or CallCompletionCancel(). If
there is currently a pending request, then the dialplan would cancel it.
This also has the advantage of showing the true state of a request, which
is an asynchronous call, even when CallCompletionRequest() thinks it was
successful. The actual request could ultimately fail. Once lit, further
feedback can be provided to the caller about the current state of their
request since it will be updated by the CCSS State Machine as appropriate.
The DEVICE_STATE mapping is configurable since the BLF being used on a
given phone type may vary. The idea is to allow some level of
customization as to the phone's behavior.
As an example, you may want the BLF key to go solid once you have
requested a callback. You may then want the LED to blink (typically
ringing) when either the callback is in process, which is a visual
indication that the incoming call is the desired callback. You may want
it to blink when the callee is ready but you are busy, giving you a visual
indication that the target is available as you may want to get off the
line so that the callback can be successful.
Device state information is sent back via the ast_devstate_prov_add()
callback for any generic CCSS device as it traverses through the state
machine. You simply provide a map between CC_STATE values and the
corresponding AST_DEVICE state values.
You could then generate hints against these states similar to what is
possible today with Custom Devstates or MeetMe states. For example, you
may have an extension 3000 that is currently associated with device
SIP/3000. You could then create a feature code for that extension that
may look something like:
exten => *823000,hint,ccss:sip/3000
You would then subscribe a BLF button to *823000 which would point to the
dialplan that handled CCSS requests/cancels using the available
DEVICE_STATE() information about ccss:sip/3000 to make the decision about
what to do.
(closes issue #18788)
Reported by: p_lindheimer
Patches:
ccss.trunk.18788.patch uploaded by p lindheimer (license 558)
Modified with final reviewboard comments.
Tested by: p_lindheimer, loloski
Review: https://reviewboard.asterisk.org/r/1105/
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r313588 | rmudgett | 2011-04-13 11:31:50 -0500 (Wed, 13 Apr 2011) | 55 lines
Merged revisions 313579 via svnmerge from
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r313579 | rmudgett | 2011-04-13 11:29:49 -0500 (Wed, 13 Apr 2011) | 48 lines
Merged revisions 313545 via svnmerge from
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r313545 | rmudgett | 2011-04-13 11:21:24 -0500 (Wed, 13 Apr 2011) | 41 lines
Asterisk does not hangup a channel after endpoint hangs up.
If the call that the dialplan started an AGI script for is hungup while
the AGI script is in the middle of a command then the AGI script is not
notified of the hangup. There are many AGI Exec commands that this can
happen with. The reported applications have been: Background, Wait, Read,
and Dial. Also the AGI Get Data command.
* Don't wait on the Asterisk channel after it has hung up. The channel is
likely to never need servicing again.
* Restored the AGI script's ability to return the AGI_RESULT_HANGUP value
in run_agi(). It previously only could return AGI_RESULT_SUCCESS or
AGI_RESULT_FAILURE after the DeadAGI and AGI applications were merged.
(closes issue #17954)
Reported by: mn3250
Patches:
issue17954_v1.8.patch uploaded by rmudgett (license 664)
issue17954_v1.6.2.patch uploaded by rmudgett (license 664)
issue17954_v1.4.patch uploaded by rmudgett (license 664)
Tested by: rmudgett
JIRA SWP-2171
(closes issue #18492)
Reported by: devmod
Tested by: rmudgett
JIRA SWP-2761
(closes issue #18935)
Reported by: nvitaly
Tested by: astmiv, rmudgett
JIRA SWP-3216
(closes issue #17393)
Reported by: siby
Tested by: rmudgett
JIRA SWP-2727
Review: https://reviewboard.asterisk.org/r/1165/
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r313366 | rmudgett | 2011-04-11 17:27:25 -0500 (Mon, 11 Apr 2011) | 2 lines
Added "Connected Line ID" and "Connected Line ID Name" to "core show channel" output.
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r313048 | jrose | 2011-04-07 08:35:33 -0500 (Thu, 07 Apr 2011) | 16 lines
Merged revisions 313047 via svnmerge from
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r313047 | jrose | 2011-04-07 08:23:01 -0500 (Thu, 07 Apr 2011) | 9 lines
Makes parking lots clear and rebuild properly when features reload is invoked from CLI
Before, default parkinglot in context parkedcalls with ext 700 would always be present and when reload was invoked, the previous parkinglots would not be cleared.
(closes issue #18801)
Reported by: mickecarlsson
Review: https://reviewboard.asterisk.org/r/1161/
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r312766 | mnicholson | 2011-04-05 09:14:50 -0500 (Tue, 05 Apr 2011) | 22 lines
Merged revisions 312764 via svnmerge from
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r312764 | mnicholson | 2011-04-05 09:13:07 -0500 (Tue, 05 Apr 2011) | 15 lines
Merged revisions 312761 via svnmerge from
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r312761 | mnicholson | 2011-04-05 09:10:34 -0500 (Tue, 05 Apr 2011) | 8 lines
Limit the number of unauthenticated manager sessions and also limit the time they have to authenticate.
AST-2011-005
(closes issue #18996)
Reported by: tzafrir
Tested by: mnicholson
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r312461 | rmudgett | 2011-04-01 16:31:39 -0500 (Fri, 01 Apr 2011) | 25 lines
CallCompletionRequest()/CallCompletionCancel() exit non-zero if fail.
The CallCompletionRequest()/CallCompletionCancel() dialplan applications
exit nonzero on normal failure conditions. The nonzero exit causes the
dialplan to hangup immediately. The dialplan author has no opportunity to
report success/failure to the user.
* Made always return zero so the dialplan can continue.
* Made set CC_REQUEST_RESULT/CC_REQUEST_REASON and
CC_CANCEL_RESULT/CC_CANCEL_REASON channel variables respectively. Also
documented the values set.
* Reduced the warning about no core instance in CallCompletionCancel() to
a debug message. It is a normal event and should not be output at the
WARNING level.
(closes issue #18763)
Reported by: p_lindheimer
Patches:
ccss.patch uploaded by p lindheimer (license 558) Modified
Tested by: p_lindheimer, rmudgett
JIRA SWP-3042
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In chan_dahdi.conf, the user can now use length 4 patterns in addition to the usual length 2 patterns. The s
ntax remains the same and the method used to track the pattern history will only change when using the length
4 patterns.
(closes issue SWP-3250)
Code:
jrose
rmudgett
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r312286 | tilghman | 2011-04-01 05:44:33 -0500 (Fri, 01 Apr 2011) | 2 lines
Reload must react correctly against a possibly changed table, so dropping the conditional reload flag.
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r312288 | tilghman | 2011-04-01 05:58:45 -0500 (Fri, 01 Apr 2011) | 21 lines
Merged revisions 312287 via svnmerge from
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r312287 | tilghman | 2011-04-01 05:51:24 -0500 (Fri, 01 Apr 2011) | 14 lines
Merged revisions 312285 via svnmerge from
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r312285 | tilghman | 2011-04-01 05:36:42 -0500 (Fri, 01 Apr 2011) | 7 lines
Found some leaking file descriptors while looking at ast_FD_SETSIZE dead code.
(issue #18969)
Reported by: oej
Patches:
20110315__issue18969__14.diff.txt uploaded by tilghman (license 14)
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r311141 | mnicholson | 2011-03-17 10:00:33 -0500 (Thu, 17 Mar 2011) | 11 lines
Merged revisions 311140 via svnmerge from
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r311140 | mnicholson | 2011-03-17 09:58:52 -0500 (Thu, 17 Mar 2011) | 4 lines
Don't write items to the manager socket twice.
AST-2011-003
(closes issue 0018987)
Reported by: ks-steven
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r310999 | twilson | 2011-03-16 14:47:59 -0500 (Wed, 16 Mar 2011) | 18 lines
Merged revisions 310998 via svnmerge from
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r310998 | twilson | 2011-03-16 14:46:36 -0500 (Wed, 16 Mar 2011) | 11 lines
Fix crash on fdopen failure
See security advisory AST-2011-004
(closes issue #18845)
Reported by: cmaj
Patches:
patch-main-tcptls-1.8.3-rc2-open-session-crash-take2.diff.txt uploaded by cmaj (license 830)
patch-main-tcptls-1.8.3-rc2-open-session-crash-take3.diff.txt uploaded by cmaj (license 830)
Tested by: cmaj, twilson
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r310993 | twilson | 2011-03-16 14:26:57 -0500 (Wed, 16 Mar 2011) | 11 lines
Merged revisions 310992 via svnmerge from
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r310992 | twilson | 2011-03-16 14:23:03 -0500 (Wed, 16 Mar 2011) | 4 lines
Don't keep trying to write to a closed connection
See security advisory AST-2011-003.
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r310902 | twilson | 2011-03-16 12:19:57 -0500 (Wed, 16 Mar 2011) | 43 lines
Merged revisions 310889 via svnmerge from
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r310889 | twilson | 2011-03-16 12:03:27 -0500 (Wed, 16 Mar 2011) | 36 lines
Merged revisions 310888 via svnmerge from
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r310888 | twilson | 2011-03-16 11:58:42 -0500 (Wed, 16 Mar 2011) | 29 lines
Don't delay DTMF in core bridge while listening for DTMF features
This patch is mostly the work of Olle Johansson. I did some cleanup and
added the silence generating code if transmit_silence is set.
When a channel listens for DTMF in the core bridge, the outbound DTMF is not
sent until we have received DTMF_END. For a long DTMF, this is a disaster. We
send 4 seconds of DTMF to Asterisk, which sends no audio for those 4 seconds.
Some products see this delay and the time skew on RTP packets that results and
start ignoring the audio that is sent afterward.
With this change, the DTMF_BEGIN frame is inspected and checked. If it matches
a feature code, we wait for DTMF_END and activate the feature as before. If
transmit_silence=yes in asterisk.conf, silence is sent if we paritally match a
multi-digit feature. If it doesn't match a feature, the frame is forwarded
along with the DTMF_END without delay. By doing it this way, DTMF is not delayed.
(closes issue #15642)
Reported by: jasonshugart
Patches:
issue_15652_dtmf_ast-1.4.patch.txt uploaded by twilson (license 396)
Tested by: globalnetinc, jde
(closes issue #16625)
Reported by: sharvanek
Review: https://reviewboard.asterisk.org/r/1092/
Review: https://reviewboard.asterisk.org/r/1125/
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r310781 | alecdavis | 2011-03-15 14:00:55 +1300 (Tue, 15 Mar 2011) | 10 lines
core show locks: display ThreadID in hexadecimal
Allow easier cross referencing of thread ID's with GDB backtraces
(closes issue #18968)
Reported by: alecdavis
Patches:
bug18968.diff.txt uploaded by alecdavis (license 585)
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r310636 | rmudgett | 2011-03-14 11:50:59 -0500 (Mon, 14 Mar 2011) | 39 lines
Merged revisions 310635 via svnmerge from
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r310635 | rmudgett | 2011-03-14 11:47:54 -0500 (Mon, 14 Mar 2011) | 32 lines
Merged revisions 310633 via svnmerge from
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r310633 | rmudgett | 2011-03-14 11:38:24 -0500 (Mon, 14 Mar 2011) | 25 lines
"Caller*ID failed checksum" on Wildcard TDM2400P and TDM410
The last character in the caller id message is getting a framing error.
The checksum is the last character in the message. A framing error in the
checksum could be because:
1) The sender did not send a full stop bit.
2) The sender cut off the FSK carrier too soon.
3) The sender opted to send zero of the specified zero to 10 trailing mark
bits and round-off errors in the code resulted in the code not being where
it thought it was in the demodulated bit stream.
Bit 8 of 'b' is set when parity error.
Bit 9 of 'b' is set when framing error.
Made ignore the framing and parity error bits if the errored character is
the checksum. We can tolerate a framing/parity error there. The checksum
character validates the message.
(closes issue #18474)
Reported by: nivek
Patches:
callerid.c.1.patch uploaded by nivek (license 636) (with modifications)
Tested by: nivek
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various OS distributions. Thanks David.
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r310287 | alecdavis | 2011-03-11 19:47:44 +1300 (Fri, 11 Mar 2011) | 17 lines
remote_bridge_loop: prevent segfault when after transfer of IAX2 of DAHDI call
If the channel condition is one of the following after breaking out of the loop, don't try to update_peer
(where x = 0/1)
1). ZOMBIE
2). cx->tech_pvt != pvtx
3). gluex != ast_rtp_instance_get_glue(cx->tech->type))
(closes issue #18781)
Reported by: alecdavis
Patches:
bug18781.diff3.txt uploaded by alecdavis (license 585)
Tested by: alecdavis, ZX81
Review: https://reviewboard.asterisk.org/r/1128/
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r310240 | twilson | 2011-03-10 10:05:45 -0600 (Thu, 10 Mar 2011) | 13 lines
Add \r\n to remaining http headers passed to ast_http_send
r309204 changed the behavior of ast_http_send. It now requires headers
to be passed with trailing \r\n. This change updates the remaining
instances in the code that did not pass the \r\n.
(closes issue #18186)
Reported by: nivaldomjunior
Patches:
res_phoneprov.c.diff uploaded by lathama (license 1028)
manager.diff.txt uploaded by twilson (license 396)
Tested by: lathama
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r309808 | tilghman | 2011-03-06 18:54:42 -0600 (Sun, 06 Mar 2011) | 14 lines
Merged revisions 309251 via svnmerge from
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r309251 | tilghman | 2011-03-01 19:06:02 -0600 (Tue, 01 Mar 2011) | 7 lines
Revert previous 2 commits, and instead conditionally redefine the same macro used in flex 2.5.35 that clashed with our workaround.
Not surprisingly, the workaround was exactly the same code as was provided by
the Flex maintainers, albeit in two different places, in different macros.
This should fix the FreeBSD builds, which have an older version of Flex.
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r309678 | tilghman | 2011-03-05 04:29:30 -0600 (Sat, 05 Mar 2011) | 14 lines
Merged revisions 309677 via svnmerge from
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r309677 | tilghman | 2011-03-05 04:28:24 -0600 (Sat, 05 Mar 2011) | 7 lines
Missed part of the conversion when we started passing ppid to astcanary.
(closes issue #18850)
Reported by: viraptor
Patches:
canary_ppid.patch uploaded by viraptor (license 543)
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(closes issue #18226)
Reported by: clegall_proformatique
Patches:
asterisk_1.8_293157_hanguprequests.svn.patch uploaded by clegall proformatique (license 1139)
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r309204 | qwell | 2011-03-01 16:25:44 -0600 (Tue, 01 Mar 2011) | 7 lines
Fix consistency of CRLFs on HTTP headers that get sent out.
(closes issue #18186)
Reported by: nivaldomjunior
Patches:
18186-httpheadernewline.diff uploaded by qwell (license 4)
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r309035 | tilghman | 2011-02-28 05:10:28 -0600 (Mon, 28 Feb 2011) | 15 lines
Merged revisions 309033-309034 via svnmerge from
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r309033 | tilghman | 2011-02-28 04:43:12 -0600 (Mon, 28 Feb 2011) | 4 lines
A later version of flex already includes the fwrite workaround code, which if used twice causes a compilation error.
Detect whether Flex will compile without the workaround; if so, suppress our workaround code.
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r309034 | tilghman | 2011-02-28 05:07:52 -0600 (Mon, 28 Feb 2011) | 2 lines
Clarify meaning, removing double negative (stupid!)
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r308903 | rmudgett | 2011-02-24 15:38:41 -0600 (Thu, 24 Feb 2011) | 9 lines
Invalid read in ast_channel_set_caller_event().
Valgrind reported that ast_channel_set_caller_event() was reading data
from a freed buffer when using the pre_set structure.
Rearange things to pre-calculate the name and number pointer before
updating the caller party structure to see if the name or number was
changed.
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r308815 | twilson | 2011-02-24 11:57:18 -0600 (Thu, 24 Feb 2011) | 26 lines
Merged revisions 308814 via svnmerge from
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r308814 | twilson | 2011-02-24 11:54:49 -0600 (Thu, 24 Feb 2011) | 19 lines
Merged revisions 308813 via svnmerge from
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r308813 | twilson | 2011-02-24 11:42:16 -0600 (Thu, 24 Feb 2011) | 12 lines
Don't broadcast FullyBooted to every AMI connection
The FullyBooted event should not be sent to every AMI connection every
time someone connects via AMI. It should only be sent to the user who
just connected.
(closes issue #18168)
Reported by: FeyFre
Patches:
bug0018168.patch uploaded by FeyFre (license 1142)
Tested by: FeyFre, twilson
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r308723 | mnicholson | 2011-02-24 09:06:14 -0600 (Thu, 24 Feb 2011) | 16 lines
Merged revisions 308722 via svnmerge from
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r308722 | mnicholson | 2011-02-24 08:59:41 -0600 (Thu, 24 Feb 2011) | 9 lines
Merged revisions 308721 via svnmerge from
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r308721 | mnicholson | 2011-02-24 08:54:56 -0600 (Thu, 24 Feb 2011) | 2 lines
silence gcc 4.2 compiler warning
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audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
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Guessed the log levels based on info that level 3
is the soft roof. Can we create a page / document
to define the levels?
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r308416 | mnicholson | 2011-02-21 09:02:20 -0600 (Mon, 21 Feb 2011) | 19 lines
Merged revisions 308414 via svnmerge from
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r308414 | mnicholson | 2011-02-21 09:00:22 -0600 (Mon, 21 Feb 2011) | 12 lines
Merged revisions 308413 via svnmerge from
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r308413 | mnicholson | 2011-02-21 08:57:15 -0600 (Mon, 21 Feb 2011) | 5 lines
Properly check the bounds of arrays when decoding UDPTL packets. Also, remove broken support for receiving UDPTL packets larger than 16k. That shouldn't ever happen anyway.
AST-2011-002
FAX-281
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enable reporting of the request URI / URL in debugging
change funny debug note to a serious note.
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The callback handle_statechange (pbx.c) fails to release its data
pointer, leaking memory in the process.
Reported by: tzafrir
Patches:
18735_pbx_free_callback.diff uploaded by tzafrir (license 46)
Review: https://reviewboard.asterisk.org/r/1110/
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Modern browsers are checking for the MIME Type of pages
and in some cases will not load a file if the type is
wrong.
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