Age | Commit message (Collapse) | Author |
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configuration files.
This option can be used to enable #exec support in the asterisk.conf configuration file.
(closes issue #16260)
Reported by: atis
Patches:
exec_includes.patch uploaded by atis (license 242)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232510 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r232444 | dvossel | 2009-12-02 13:03:01 -0600 (Wed, 02 Dec 2009) | 9 lines
fixes app_queue ao2 error
(closes issue #16369)
Reported by: vrban
Patches:
queue_issue_1.4.diff uploaded by dvossel (license 671)
Tested by: dvossel
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terminate recording.
(closes issue #15436)
Reported by: Vince
Patches:
app_record.diff uploaded by Vince (license 823)
Tested by: dbrooks
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232442 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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on an inbound sip_pvt if the outbound call is redirected.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r232355 | file | 2009-12-02 13:04:52 -0400 (Wed, 02 Dec 2009) | 5 lines
Fix a bug where if you hung up very quickly after calling AMD it would overwrite the AMDSTATUS of HANGUP with TOOLONG.
(closes issue #16239)
Reported by: CGMChris
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r232350 | dvossel | 2009-12-02 10:59:18 -0600 (Wed, 02 Dec 2009) | 6 lines
ast_outaddrfor doesn't do htons() on port, looks odd in strace.
(closes issue #16290)
Reported by: wdoekes
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for a 488 Not acceptable here response.
(closes issue #16186)
Reported by: atis
Patches:
sip_t38_response_415.patch uploaded by atis (license 242)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232345 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r232268 | dvossel | 2009-12-02 09:41:36 -0600 (Wed, 02 Dec 2009) | 9 lines
fixes segfault in func_groupcount
closes issue #16337)
Reported by: Parantido
Patches:
issue_16337.diff uploaded by dvossel (license 671)
Tested by: Parantido, dvossel
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certain scenario
causing code to execute during reload that should not.
(issue AST-263)
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r232165 | twilson | 2009-12-01 22:05:09 -0600 (Tue, 01 Dec 2009) | 2 lines
Fix compiling without devmode
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The manpage lies; it says that the function is in POSIX, but that's only for
ffs(3), not ffsll(3).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232164 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r232090 | jpeeler | 2009-12-01 18:42:58 -0600 (Tue, 01 Dec 2009) | 10 lines
Do not modify the gain settings on data calls.
(The digital flag actually represents a data call.)
(closes issue #15972)
Reported by: udosw
Patches:
transcap_digital_fix.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r232007 | russell | 2009-12-01 17:25:36 -0600 (Tue, 01 Dec 2009) | 2 lines
Fix a warning pointed out by buildbot.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r231911 | jpeeler | 2009-12-01 15:29:31 -0600 (Tue, 01 Dec 2009) | 12 lines
Fix crash with invalid frame data
The crash was happening as a result of a frame containing an invalid data
pointer, but was set with data length of zero. The few times the issue was
reproduced it _seemed_ that the frame was queued properly, that is the data
pointer was set to NULL. I never could reproduce the crash so as a last resort
the crash has been fixed, but a check in __ast_read has been added to give as
much information about the source of problematic frames in the future.
(closes issue #16058)
Reported by: atis
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r231853 | dvossel | 2009-12-01 15:14:31 -0600 (Tue, 01 Dec 2009) | 3 lines
WaitExten m option with no parameters generates frame with zero datalen but non-null data ptr
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In the process of swapping ULAW to a place in the extended codec space, we
found several unhandled cases, where a 32-bit integer was still being used to
handle a codec field. Most of these have been fixed with this commit, although
there is at least one case (codec_dahdi) which depends upon outside headers to
be altered before a conversion can be made.
(Fixes AST-278, SWP-459)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r231740 | mnicholson | 2009-12-01 09:34:57 -0600 (Tue, 01 Dec 2009) | 2 lines
Ignore unknown formats in ast_format_str_reduce() and return an error if no know formats are found.
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1) Allow users of UDPTL stack to associate a character-string tag with a UDPTL
session, so that log/error/debug messages generated by the UDPTL stack can
be 'connected' to the endpoint that caused them to be generated.
2) Improve comments (and process) of calculating the far end's maximum IFP size
when redundancy mode is in use for error correction.
3) When an IFP larger than the calculated 'far max IFP' size is presented for
writing, truncate it rather than putting in the buffer and allowing the buffer
to overflow; this will cause the ends to retrain to a lower bit rate that
produces IFPs of an appropriate size if possible, and if not possible, the
FAX transfer will fail completely. In these cases, it is due to the one endpoint
supplying a T38FaxMaxDatagram value that is improperly calculated and is
too low to be of use; we have configuration options available to override
this behavior.
4) Eliminate use of T38FaxMaxDatagram value in udptl.conf; it is no longer
needed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231692 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov 2009) | 8 lines
Remove duplicate entries from voicemail format lists. This prevents app_voicemail from entering an infinite loop when the same format is specified twice in the format list.
(closes issue #15625)
Reported by: Shagg63
Tested by: mnicholson
Review: https://reviewboard.asterisk.org/r/429/
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov 2009) | 8 lines
Remove duplicate entries from voicemail format lists. This prevents app_voicemail from entering an infinite loop when the same format is specified twice in the format list.
(closes issue #15625)
Reported by: Shagg63
Tested by: mnicholson
Review: https://reviewboard.asterisk.org/r/429/
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as a fatal error.
(closes issue #16238)
Reported by: seandarcy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231602 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch adds a ref to the queue_ent object's parent call_queue
in queue_exec() so the call_queue won't be destroyed
while the the queue_ent still holds a pointer to it.
(closes issue 0015686)
Tested by: dvossel, aragon
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231556 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r231441 | dvossel | 2009-11-30 11:14:08 -0600 (Mon, 30 Nov 2009) | 11 lines
fixes crash caused by RTP comfort noise payload greater than 24 bytes
AST-2009-010
(closes issue #16242)
Reported by: amorsen
Patches:
issue16242.diff uploaded by oej (license 306)
Tested by: amorsen, oej, dvossel
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(closes issue #15193)
Reported by: eliel
Patches:
20091111__issue15193.diff.txt uploaded by tilghman (license 14)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231439 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r231437 | dvossel | 2009-11-30 10:32:58 -0600 (Mon, 30 Nov 2009) | 18 lines
app_queue crashes randomly, often during call-transfers
In app_queue, it is possible for a call_queue to be destroyed
while another object still holds a pointer to it. This patch
converts call_queue objects to ao2 objects allowing them to be
ref counted. This makes it safe for the queue_ent object in
queue_exec() to reference it's parent call_queue even after it
has left the queue.
(closes issue #15686)
Reported by: Hatrix
Patches:
v2_queue_ao2.diff uploaded by dvossel (license 671)
Tested by: dvossel, aragon
Review: https://reviewboard.asterisk.org/r/427/
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get lost. Now we just
indicate unhold directly after the masquerade is complete.
(issue ABE-2011)
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2. Added destination port support.
3. Added new protocols.
4. Added QoS.
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with guidelines
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r231298 | tilghman | 2009-11-25 16:31:57 -0600 (Wed, 25 Nov 2009) | 2 lines
After a frame duplication failure, unlock the channel before returning.
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r231235 | dvossel | 2009-11-25 15:38:32 -0600 (Wed, 25 Nov 2009) | 9 lines
fixes solaris segfault on dial with verbosity >= 3
(closes issue #16193)
Reported by: asgaroth
Patches:
bug_16193_1.4.21.2_vers.diff uploaded by snuffy (license 35)
Tested by: asgaroth, snuffy
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r231233 | dvossel | 2009-11-25 15:23:41 -0600 (Wed, 25 Nov 2009) | 8 lines
fixes conditional jump or move depending on uninitialised STACK value
(closes issue #16261)
Reported by: edguy3
Patches:
edguy16261.patch uploaded by edguy3 (license 917)
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Found by Maxim Litnitskiy.
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debug statements.
(closes issue #15982, closes issue #15984)
Reported by: atis
Patches:
20091111__issue15982.diff.txt uploaded by tilghman (license 14)
Tested by: atis
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231134 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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ast_spawn_extension behaves differently from 1.4 in that hangups and extensions
that do not exist do not return an error, whereas in 1.6 it does. This is now
taken into account so that the AST_FLAG_BRIDGE_HANGUP_RUN flag gets set
properly.
(closes issue #16106)
Reported by: ajohnson
Tested by: ajohnson
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Changed areas in sig_pri to set the digital flag using a callback that will
also set the corresponding flag in chan_dahdi. Modified dahdi_request slightly
so that if a bearer is marked as digital, that information is available when
creating the new channel.
(closes issue #16151)
Reported by: alecdavis
Patch based on bug_16151.diff.txt uploaded by alecdavis (license 585)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231058 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r230964
(related to issue #14155)
Reported by: junky
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the ast_get_encoded_* functions.
* Add REPLACE function, which searches a given variable for a set of
characters and replaces each with a given character.
* Add PASSTHRU function, which passes a literal string back, like a NoOp for
functions. Intent is to be able to specify a literal string to another
function that takes a variable name as an argument.
* Let the array manipulation functions work with dialplan functions, in
addition to variables. This allows the array manipulation functions to
modify ASTDB and ODBC backends, assuming the func_odbc configuration has
both read and write functions.
(closes issue #15223)
Reported by: ajohnson
Patches:
20091112__issue15223.diff.txt uploaded by tilghman (license 14)
Tested by: lmadsen, tilghman
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(closes issue #14155)
Reported by: junky
Patches:
M14555-trunk.diff uploaded by junky (license 177) (modified)
Tested by: junky
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Internally the way T.38 is negotiated has changed and the option no longer
reflects a behavior that is valid. It will now look for a CNG tone on
received calls and if present send the call to the 'fax' extension. It is
then up to the application or channel to request the switch over to T.38.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r230839 | kpfleming | 2009-11-23 09:09:24 -0600 (Mon, 23 Nov 2009) | 1 line
Correct fix for issue #16268... the reporter's original patch was very close to correct.
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r230875 | kpfleming | 2009-11-23 09:31:02 -0600 (Mon, 23 Nov 2009) | 7 lines
When 'sip set debug' is enabled, and the last line of an incoming SIP message
is not properly newline terminated, ensure that that line is included in the
debug output.
(part of issue #16268)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r230772 | kpfleming | 2009-11-23 08:13:56 -0600 (Mon, 23 Nov 2009) | 5 lines
Ensure that SDP parsing does not ignore the last line of the SDP.
(closes issue #16268)
Reported by: sgimeno
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(closes issue #16094)
Reported by: alecdavis
Patches:
bug16094.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis, dvossel
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original problem, while gcc investigates.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r230627 | mnicholson | 2009-11-20 14:53:06 -0600 (Fri, 20 Nov 2009) | 8 lines
Copy the peer CDR's userfield to the bridge CDR if it exists. This is necessary for the recordagentcalls option in chan_agent to store the recorded file name in the bridge CDR.
(closes issue #14590)
Reported by: msetim
Patches:
queue_agent_userfield.patch uploaded by Laureano (license 265)
Tested by: Laureano, mnicholson
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also include a better cmd define for S command
Review: https://reviewboard.asterisk.org/r/430/
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