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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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@209279 f38db490-d61c-443f-a65b-d21fe96a405b
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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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@209256 f38db490-d61c-443f-a65b-d21fe96a405b
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"initialize"
(closes issue #15571)
Reported by: alecdavis
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@209098 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@208593 f38db490-d61c-443f-a65b-d21fe96a405b
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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/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@208464 f38db490-d61c-443f-a65b-d21fe96a405b
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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
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@208113 f38db490-d61c-443f-a65b-d21fe96a405b
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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
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@208017 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@207551 f38db490-d61c-443f-a65b-d21fe96a405b
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This helps to prevent a crash which may occur due to our freeing
garbage due to a struct being uninitialized.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@207522 f38db490-d61c-443f-a65b-d21fe96a405b
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Reported by "Hoggins!" on asterisk-dev list.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@207317 f38db490-d61c-443f-a65b-d21fe96a405b
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lengths more consistent.
(closes issue #15493)
Reported by: lasko
Patches:
meetme.diff uploaded by lasko (license 833)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@206567 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@206490 f38db490-d61c-443f-a65b-d21fe96a405b
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If a redirecting control frame is processed or a call forward occurs,
we need to reset the sentringing flag so that we can send another ringing
indication to the phone that may contain a connected line update.
AST-164
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@206455 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #15427)
Reported by: brushtyler
Patches:
app_voicemail.c.diff uploaded by brushtyler (license 821)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@206185 f38db490-d61c-443f-a65b-d21fe96a405b
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The transmit_audio() and transmit_t38() functions in app_fax have processing
loops that are supposed to wait for frames to arrive on the channel and then
handle them, but they also have short timeouts so that the loops can have
watchdog timers and do other required processing. This commit changes the loops
to not actually call ast_read() and attempt to process the returned frame
unless a frame actually arrived, eliminating hundreds of LOG_DEBUG messages
and slightly improving performance.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@205780 f38db490-d61c-443f-a65b-d21fe96a405b
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Revision 205696 did not quite fix all the issues with the T.38 negotiation
changes and app_fax; this patch corrects them, along with a couple of other
minor issues.
(closes issue #15480)
Reported by: dimas
Patches:
test2-15480.patch uploaded by dimas (license 88)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@205770 f38db490-d61c-443f-a65b-d21fe96a405b
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Recent changes in T.38 negotiation in Asterisk caused these applications to
not respond when the other endpoint initiated a switchover to T.38; this
resulted in the T.38 switchover failing, and the FAX attempt to be made
using an audio connection, instead of T.38 (which would usually cause the
FAX to fail completely).
This patch corrects this problem, and the applications will now correctly
respond to the T.38 switchover request. In addition, the response will include
the appopriate T.38 session parameters based on what the other end offered
and what our end is capable of.
(closes issue #14849)
Reported by: afosorio
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@205696 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r205349 | mmichelson | 2009-07-08 14:26:13 -0500 (Wed, 08 Jul 2009) | 14 lines
Prevent phantom calls to queue members.
If a caller were to hang up while a periodic announcement or position
were being said, the return value for those functions would incorrectly
indicate that the caller was still in the queue. With these changes,
the problem does not occur.
(closes issue #14631)
Reported by: latinsud
Patches:
queue_announce_ghost_call2.diff uploaded by latinsud (license 745)
(with small modification from me)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@205350 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@204622 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@204561 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r204469 | tilghman | 2009-06-30 13:23:35 -0500 (Tue, 30 Jun 2009) | 11 lines
"tw" is the language specification for Twi (from Ghana) not Taiwanese.
(closes issue #15346)
Reported by: volivier
Patches:
20090617__issue15346__1.4.diff.txt uploaded by tilghman (license 14)
20090617__issue15346__trunk.diff.txt uploaded by tilghman (license 14)
20090617__issue15346__1.6.0.diff.txt uploaded by tilghman (license 14)
20090617__issue15346__1.6.1.diff.txt uploaded by tilghman (license 14)
20090617__issue15346__1.6.2.diff.txt uploaded by tilghman (license 14)
Tested by: volivier
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@204470 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@204355 f38db490-d61c-443f-a65b-d21fe96a405b
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Also, the code in this module is horrendous and we should remove it from the
tree. I'm not sure who is supposed to be maintaning this thing, but they
clearly are not. I don't see the sense of leaving it in the main tree. If it
lives *anywhere* it should be in addons.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@204143 f38db490-d61c-443f-a65b-d21fe96a405b
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left to spy on.
(closes issue #14594)
Reported by: JimDickenson
Patches:
chanspy.diff uploaded by JimDickenson (license 710)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@203842 f38db490-d61c-443f-a65b-d21fe96a405b
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Max silence was represented in milliseconds, yet vmminsecs (minmessage) was represented
as seconds.
Also, the inequality was reversed. The warning, if triggered, was "Max silence should
be less than minmessage or you may get empty messages", which should have been logged
if max silence was greater than minmessage, but the check was for less than.
Also, conforming if statement to coding guidelines.
closes issue #15331)
Reported by: markd
Review: https://reviewboard.asterisk.org/r/293/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@203721 f38db490-d61c-443f-a65b-d21fe96a405b
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application and channel.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@203699 f38db490-d61c-443f-a65b-d21fe96a405b
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CEL is the new system for logging channel events. This was inspired after
facing many problems trying to represent what is possible to happen to a call
in Asterisk using CDR records. For more information on CEL, see the built in
HTML or PDF documentation generated from the files in doc/tex/.
Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard
work developing this code. Also, thanks to Matt Nicholson (mnicholson) and
Sean Bright (seanbright) for their assistance in the final push to get this
code ready for Asterisk trunk.
Review: https://reviewboard.asterisk.org/r/239/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@203638 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #15355)
Reported by: deuffy
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@202183 f38db490-d61c-443f-a65b-d21fe96a405b
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within app_voicemail for directory functions. It is therefore no longer
necessary for app_directory to be linked against the ODBC libraries (and it
never was necessary for app_directory to be linked against IMAP, though it
was).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@201783 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #15269)
Reported by: contactmayankjain
Patches:
patch.txt uploaded by contactmayankjain (license 740)
memory_leak_stuff.trunk.diff uploaded by dvossel (license 671)
Tested by: contactmayankjain, dvossel
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@201678 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #15186)
Reported by: ajohnson
Patches:
20090528__issue15186.diff.txt uploaded by tilghman (license 14)
Tested by: ajohnson
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@201531 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r201423 | dvossel | 2009-06-17 14:28:12 -0500 (Wed, 17 Jun 2009) | 19 lines
StopMixMonitor race condition (not giving up file immediately)
StopMixMonitor only indicates to the MixMonitor thread to stop
writing to the file. It does not guarantee that the recording's
file handle is available to the dialplan immediately after execution.
This results in a race condition. To resolve this, the filestream
pointer is placed in a datastore on the channel. When StopMixMonitor
is called, the datastore is retrieved from the channel and the
filestream is closed immediately before returning to the dialplan.
Documentation indicating the use of StopMixMonitor to free files
has been updated as well.
(closes issue #15259)
Reported by: travisghansen
Tested by: dvossel
Review: https://reviewboard.asterisk.org/r/283/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@201445 f38db490-d61c-443f-a65b-d21fe96a405b
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Some applications (notably app_fax) do not need digit detection nor FAX tone
detection while they are running, and if Asterisk is using software DSPs to provide
the detection, this consumes extra CPU cycles that could be better spent on the
actual application. This patch allows applications to query and control the state
of digit and tone detection on a channel, and modifies app_fax to disable them
while the FAX operations are occurring (and re-enable digit detection afterwards).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@201139 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun 2009) | 11 lines
Improve support for media paths that can generate multiple frames at once.
There are various media paths in Asterisk (codec translators and UDPTL, primarily)
that can generate more than one frame to be generated when the application calling
them expects only a single frame. This patch addresses a number of those cases,
at least the primary ones to solve the known problems. In addition it removes the
broken TRACE_FRAMES support, fixes a number of bugs in various frame-related API
functions, and cleans up various code paths affected by these changes.
https://reviewboard.asterisk.org/r/175/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@201056 f38db490-d61c-443f-a65b-d21fe96a405b
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Voicemail can only use one storage module at the moment.
Because it's unclear that selecting one of the storage modules
in menuselect will disable filesystem storage we now have
a FILE_STORAGE option that conflicts with the other modules.
(closes issue #15333)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@200943 f38db490-d61c-443f-a65b-d21fe96a405b
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Fix up modules in the 'apps' directory, and also correct the bad example of
enum definitions in include/asterisk/app.h, which many developers followed
(thanks for reading the documentation!). In addition, add some basic usage
examples of the 'pahole' and 'pglobal' tools to the coding guidelines.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@200656 f38db490-d61c-443f-a65b-d21fe96a405b
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This patch provides a new implementation of the optional API support defined
in asterisk/optional_api.h; this new version provides solves compatibility
issues with the use of linker version scripts for suppressing global symbols.
In addition, there is now a functional (and tested!) implementation for Mac OS/X,
so module writers no longer need to use special tests before calling optional
API functions. All future implementations must provide these same semantics,
so that module writers can rely on them.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@200519 f38db490-d61c-443f-a65b-d21fe96a405b
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extension from a queue.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@200326 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@200290 f38db490-d61c-443f-a65b-d21fe96a405b
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Move OSP* applications static documentation to the new AstXML form.
(closes issue #15245)
Reported by: eliel
Patches:
app_osplookup_static_conversion.txt uploaded by lmadsen (license 10)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@199547 f38db490-d61c-443f-a65b-d21fe96a405b
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Move application ExternalIVR static documentation to the new AstXML form.
(issue #15245)
Reported by: eliel
Patches:
app_externalivr.diff uploaded by eliel (license 64)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@199514 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@199479 f38db490-d61c-443f-a65b-d21fe96a405b
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Move AGI command 'gosub' statis documentation to the new AstXML form.
(issue #15245)
Reported by: eliel
Patches:
app_stack_static_conversion.txt uploaded by lmadsen (license 10)
(with minor changes by me)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@199446 f38db490-d61c-443f-a65b-d21fe96a405b
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Move function MEETME_INFO static documentation to the new AstXML form.
(issue #15245)
Reported by: eliel
Patches:
app_meetme_static_conversion.txt uploaded by lmadsen (license 10)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@199409 f38db490-d61c-443f-a65b-d21fe96a405b
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Move function MINIVMACCOUNT and MINIVMCOUNTER statis documentation to the new
AstXML form.
(issue #15245)
Reported by: eliel
Patches:
app_minivm_static_conversion.txt uploaded by lmadsen (license 10)
(with minor changes by me)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@199376 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@199372 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@199370 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@198954 f38db490-d61c-443f-a65b-d21fe96a405b
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When connected line updates are received or generated in the middle
of an application call, it is now possible to execute a macro to
manipulate the connected line data. This way, phone numbers may be
manipulated to be more presentable to users, names may be changed
for...whatever reason, or whatever else needs to be done may be.
Review: https://reviewboard.asterisk.org/r/256
AST-165
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@198727 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@198530 f38db490-d61c-443f-a65b-d21fe96a405b
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