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(closes issue #12912)
Reported by: rathaus
Tested by: tilghman, russell, dvossel, dbrooks
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r210908 | tilghman | 2009-08-06 16:29:26 -0500 (Thu, 06 Aug 2009) | 9 lines
Allow Gosub to recognize quote delimiters without consuming them.
(closes issue #15557)
Reported by: rain
Patches:
20090723__issue15557.diff.txt uploaded by tilghman (license 14)
Tested by: rain
Review: https://reviewboard.asterisk.org/r/316/
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r209554 | dbrooks | 2009-07-30 11:07:05 -0500 (Thu, 30 Jul 2009) | 6 lines
Fixes numerous spelling errors. Patch submitted by alecdavis.
(closes issue #15595)
Reported by: alecdavis
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r209098 | dbrooks | 2009-07-27 11:33:50 -0500 (Mon, 27 Jul 2009) | 6 lines
Fixing typos. Replaces "recieved" with "received" and "initilize" with "initialize"
(closes issue #15571)
Reported by: alecdavis
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r208548 | kpfleming | 2009-07-24 10:02:53 -0500 (Fri, 24 Jul 2009) | 8 lines
Resolve a T.38 negotiation issue left over from the udptl-updates merge.
The udptl-updates branch that was merged yesterday failed to properly send back
T.38 SDP responses with the correct error correction mode, if the incoming SDP
from the other end caused us to change error correction modes. This patch
corrects that situation.
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r208464 | kpfleming | 2009-07-23 16:57:24 -0500 (Thu, 23 Jul 2009) | 46 lines
Rework of T.38 negotiation and UDPTL API to address interoperability problems
Over the past couple of months, a number of issues with Asterisk
negotiating (and successfully completing) T.38 sessions with various
endpoints have been found. This patch attempts to address many of
them, primarily focused around ensuring that the endpoints'
MaxDatagram size is honored, and in addition by ensuring that T.38
session parameter negotiation is performed correctly according to the
ITU T.38 Recommendation.
The major changes here are:
1) T.38 applications in Asterisk (app_fax) only generate/receive IFP
packets, they do not ever work with UDPTL packets. As a result of
this, they cannot be allowed to generate packets that would overflow
the other endpoints' MaxDatagram size after the UDPTL stack adds any
error correction information. With this patch, the application is told
the maximum *IFP* size it can generate, based on a calculation using
the far end MaxDatagram size and the active error correction mode on
the T.38 session. The same is true for sending *our* MaxDatagram size
to the remote endpoint; it is computed from the value that the
application says it can accept (for a single IFP packet) combined with
the active error correction mode.
2) All treatment of T.38 session parameters as 'capabilities' in
chan_sip has been removed; these parameters are not at all like
audio/video stream capabilities. There are strict rules to follow for
computing an answer to a T.38 offer, and chan_sip now follows those
rules, using the desired parameters from the application (or channel)
that wants to accept the T.38 negotiation.
3) chan_sip now stores and forwards ast_control_t38_parameters
structures for tracking 'our' and 'their' T.38 session parameters;
this greatly simplifies negotiation, especially for pass-through
calls.
4) Since T.38 negotiation without specifying parameters or receiving
the final negotiated parameters is not very worthwhile, the
AST_CONTROL_T38 control frame has been removed. A note has been added
to UPGRADE.txt about this removal, since any out-of-tree applications
that use it will no longer function properly until they are upgraded
to use AST_CONTROL_T38_PARAMETERS.
Review: https://reviewboard.asterisk.org/r/310/
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r205696 | kpfleming | 2009-07-09 16:20:23 -0500 (Thu, 09 Jul 2009) | 16 lines
Repair ability of SendFAX/ReceiveFAX to respond to T.38 switchover.
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
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r205600 | dvossel | 2009-07-09 11:19:09 -0500 (Thu, 09 Jul 2009) | 9 lines
Merged revisions 205599 via svnmerge from
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r205599 | dvossel | 2009-07-09 11:18:09 -0500 (Thu, 09 Jul 2009) | 2 lines
Changing ast_samp2tv to not use floating point.
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r205479 | dvossel | 2009-07-08 18:19:09 -0500 (Wed, 08 Jul 2009) | 16 lines
Merged revisions 205471 via svnmerge from
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r205471 | dvossel | 2009-07-08 18:15:54 -0500 (Wed, 08 Jul 2009) | 10 lines
Fixes 8khz assumptions
Many calculations assume 8khz is the codec rate. This
is not always the case. This patch only addresses chan_iax.c
and res_rtp_asterisk.c, but I am sure there are other areas
that make this assumption as well.
Review: https://reviewboard.asterisk.org/r/306/
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r205412 | dvossel | 2009-07-08 17:15:06 -0500 (Wed, 08 Jul 2009) | 12 lines
Merged revisions 205409 via svnmerge from
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r205409 | dvossel | 2009-07-08 16:35:12 -0500 (Wed, 08 Jul 2009) | 6 lines
moving ast_devstate_to_extenstate to pbx.c from devicestate.c
ast_devstate_to_extenstate belongs in pbx.c. This change
fixes a compile time error with chan_vpb as well.
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r205216 | dvossel | 2009-07-08 11:54:24 -0500 (Wed, 08 Jul 2009) | 17 lines
Merged revisions 205215 via svnmerge from
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r205215 | dvossel | 2009-07-08 11:53:40 -0500 (Wed, 08 Jul 2009) | 10 lines
ast_samp2tv needs floating point for 16khz audio
In ast_samp2tv(), (1000000 / _rate) = 62.5 when _rate is 16000.
The .5 is currently stripped off because we don't calculate
using floating points. This causes madness with 16khz audio.
(issue ABE-1899)
Review: https://reviewboard.asterisk.org/r/305/
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r205120 | russell | 2009-07-08 10:17:19 -0500 (Wed, 08 Jul 2009) | 16 lines
Move OpenSSL initialization to a single place, make library usage thread-safe.
While doing some reading about OpenSSL, I noticed a couple of things that
needed to be improved with our usage of OpenSSL.
1) We had initialization of the library done in multiple modules. This has now
been moved to a core function that gets executed during Asterisk startup.
We already link OpenSSL into the core for TCP/TLS functionality, so this
was the most logical place to do it.
2) OpenSSL is not thread-safe by default. However, making it thread safe is
very easy. We just have to provide a couple of callbacks. One callback
returns a thread ID. The other handles locking. For more information,
start with the "Is OpenSSL thread-safe?" question on the FAQ page of
openssl.org.
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This patch adds reference counting for sip dialogs into 1.6.0.
When proc_session_timer() is called from the scheduler thread
it has no guarantee the session timer's dialog won't be freed
from underneath it. Now the session timer holds a reference
to the dialog, preventing it from being destroyed during the
middle of proc_session_timer().
(closes issue #13623)
Reported by: Nik Soggia
Review: https://reviewboard.asterisk.org/r/302/
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r204710 | dvossel | 2009-07-02 11:03:44 -0500 (Thu, 02 Jul 2009) | 21 lines
Merged revisions 204681 via svnmerge from
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r204681 | dvossel | 2009-07-02 10:05:57 -0500 (Thu, 02 Jul 2009) | 14 lines
Improved mapping of extension states from combined device states.
This fixes a few issues with incorrect extension states and adds
a cli command, core show device2extenstate, to display all possible
state mappings.
(closes issue #15413)
Reported by: legart
Patches:
exten_helper.diff uploaded by dvossel (license 671)
Tested by: dvossel, legart, amilcar
Review: https://reviewboard.asterisk.org/r/301/
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r203699 | file | 2009-06-26 16:27:24 -0300 (Fri, 26 Jun 2009) | 2 lines
Improve T.38 negotiation by exchanging session parameters between application and channel.
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r201262 | kpfleming | 2009-06-17 07:04:17 -0500 (Wed, 17 Jun 2009) | 15 lines
Merged revisions 201261 via svnmerge from
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r201261 | kpfleming | 2009-06-17 07:03:25 -0500 (Wed, 17 Jun 2009) | 9 lines
Correct AST_LIST_APPEND_LIST behavior when list to be appended is empty.
When the list to be appended is empty, and the list to be appended to is *not*,
AST_LIST_APPEND_LIST would actually cause the target list to become broken,
and no longer have a pointer to its last entry. This patch fixes the problem.
(reported by Stanislaw Pitucha on the asterisk-dev mailing list)
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r201056 | kpfleming | 2009-06-16 13:54:30 -0500 (Tue, 16 Jun 2009) | 18 lines
Merged revisions 200991 via svnmerge from
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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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r201090 | kpfleming | 2009-06-16 14:27:12 -0500 (Tue, 16 Jun 2009) | 5 lines
Another minor fix to compiler attribute checking.
Defaulting to 'static' for the function scope was bad... so remove it.
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r200764 | kpfleming | 2009-06-15 20:28:08 -0500 (Mon, 15 Jun 2009) | 11 lines
Ensure that configure-script testing for compiler attributes actually works.
The configure script tests for compiler attributes didn't actually enable
enough warnings or provide a proper test harness to determine whether the
compiler supports the attribute in question or not; this caused gcc 4.1 to
report that it supports 'weakref', but it doesn't actually support it in the
way that is needed for our optional API mechanism. The new configure script
test will properly distinguish between full support and partial support
for this attribute, among others.
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r199857 | seanbright | 2009-06-10 12:10:23 -0400 (Wed, 10 Jun 2009) | 9 lines
Merged revisions 199856 via svnmerge from
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r199856 | seanbright | 2009-06-10 12:08:35 -0400 (Wed, 10 Jun 2009) | 2 lines
__WORDSIZE is not available on all platforms, so use sizeof(void *) instead.
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r199630 | seanbright | 2009-06-08 15:33:09 -0400 (Mon, 08 Jun 2009) | 32 lines
Merged revisions 199626,199628 via svnmerge from
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r199626 | seanbright | 2009-06-08 15:24:32 -0400 (Mon, 08 Jun 2009) | 21 lines
Increase the size of our thread stack on 64 bit processors.
We were setting the stack size for each thread to 240KB regardless of
architecture, which meant that in some scenarios we actually had less available
stack space on 64 bit processors (pointers use 8 bytes instead of 4). So now we
calculate the stack size we reserve based on the platform's __WORDSIZE, which
gives us:
32 bit -> 240KB
64 bit -> 496KB
128 bit -> 1008KB (that's right, we're ready for 128 bit processors)
Patch typed by me but written by several members of #asterisk-dev, including
Kevin, Tilghman, and Qwell.
(closes issue #14932)
Reported by: jpiszcz
Patches:
06052009_issue14932.patch uploaded by seanbright (license 71)
Tested by: seanbright
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r199628 | seanbright | 2009-06-08 15:28:33 -0400 (Mon, 08 Jun 2009) | 2 lines
Fix a typo in the stack size calculation just introduced.
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r199051 | seanbright | 2009-06-04 10:31:24 -0400 (Thu, 04 Jun 2009) | 47 lines
Merged revisions 199022 via svnmerge from
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r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun 2009) | 40 lines
Safely handle AMI connections/reload requests that occur during startup.
During asterisk startup, a lock on the list of modules is obtained by the
primary thread while each module is initialized. Issue 13778 pointed out a
problem with this approach, however. Because the AMI is loaded before other
modules, it is possible for a module reload to be issued by a connected client
(via Action: Command), causing a deadlock.
The resolution for 13778 was to move initialization of the manager to happen
after the other modules had already been lodaded. While this fixed this
particular issue, it caused a problem for users (like FreePBX) who call AMI
scripts via an #exec in a configuration file (See issue 15189).
The solution I have come up with is to defer any reload requests that come in
until after the server is fully booted. When a call comes in to
ast_module_reload (from wherever) before we are fully booted, the request is
added to a queue of pending requests. Once we are done booting up, we then
execute these deferred requests in turn.
Note that I have tried to make this a bit more intelligent in that it will not
queue up more than 1 request for the same module to be reloaded, and if a
general reload request comes in ('module reload') the queue is flushed and we
only issue a single deferred reload for the entire system.
As for how this will impact existing installations - Before 13778, a reload
issued before module initialization was completed would result in a deadlock.
After 13778, you simply couldn't connect to the manager during startup (which
causes problems with #exec-that-calls-AMI configuration files). I believe this
is a good general purpose solution that won't negatively impact existing
installations.
(closes issue #15189)
(closes issue #13778)
Reported by: p_lindheimer
Patches:
06032009_15189_deferred_reloads.diff uploaded by seanbright (license 71)
Tested by: p_lindheimer, seanbright
Review: https://reviewboard.asterisk.org/r/272/
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r198856 | dvossel | 2009-06-02 16:17:49 -0500 (Tue, 02 Jun 2009) | 10 lines
Generic call forward api, ast_call_forward()
The function ast_call_forward() forwards a call to an extension specified in an ast_channel's call_forward string. After an ast_channel is called, if the channel's call_forward string is set this function can be used to forward the call to a new channel and terminate the original one. I have included this api call in both channel.c's ast_request_and_dial() and feature.c's feature_request_and_dial(). App_dial and app_queue already contain call forward logic specific for their application and options.
(closes issue #13630)
Reported by: festr
Review: https://reviewboard.asterisk.org/r/271/
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r198072 | mnicholson | 2009-05-29 14:04:24 -0500 (Fri, 29 May 2009) | 21 lines
Merged revisions 198068 via svnmerge from
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r198068 | mnicholson | 2009-05-29 13:53:01 -0500 (Fri, 29 May 2009) | 15 lines
Use AST_CDR_NOANSWER instead of AST_CDR_NULL as the default CDR disposition.
This change also involves the addition of an AST_CDR_FLAG_ORIGINATED flag that is used on originated channels to distinguish: them from dialed channels.
(closes issue #12946)
Reported by: meral
Patches:
null-cdr2.diff uploaded by mnicholson (license 96)
Tested by: mnicholson, dbrooks
(closes issue #15122)
Reported by: sum
Tested by: sum
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r197606 | mmichelson | 2009-05-28 10:32:19 -0500 (Thu, 28 May 2009) | 22 lines
Recorded merge of revisions 197588 via svnmerge from
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r197588 | mmichelson | 2009-05-28 10:27:49 -0500 (Thu, 28 May 2009) | 16 lines
Allow for media to arrive from an alternate source when responding to a reinvite with 491.
When we receive a SIP reinvite, it is possible that we may not be able to process the
reinvite immediately since we have also sent a reinvite out ourselves. The problem is
that whoever sent us the reinvite may have also sent a reinvite out to another party,
and that reinvite may have succeeded.
As a result, even though we are not going to accept the reinvite we just received, it
is important for us to not have problems if we suddenly start receiving RTP from a new
source. The fix for this is to grab the media source information from the SDP of the
reinvite that we receive. This information is passed to the RTP layer so that it will
know about the alternate source for media.
Review: https://reviewboard.asterisk.org/r/252
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r197543 | mmichelson | 2009-05-28 09:58:06 -0500 (Thu, 28 May 2009) | 27 lines
Merged revisions 197537 via svnmerge from
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r197537 | mmichelson | 2009-05-28 09:49:13 -0500 (Thu, 28 May 2009) | 21 lines
Add flags to chanspy audiohook so that audio stays in sync.
There are two flags being added to the chanspy audiohook here. One
is the pre-existing AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set,
we ensure that the read and write slinfactories on the audiohook do
not skew beyond a certain tolerance.
In addition, there is a new audiohook flag added here,
AST_AUDIOHOOK_SMALL_QUEUE. With this flag set, we do not allow for
a slinfactory to build up a substantial amount of audio before
flushing it. For this particular issue, this means that the person
spying on the call will hear the conversations in real time with very
little delay in the audio.
(closes issue #13745)
Reported by: geoffs
Patches:
13745.patch uploaded by mmichelson (license 60)
Tested by: snblitz
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The bridge was terminating immediately after the attended transfer was
completed. The problem was because upon reentering ast_channel_bridge
nexteventts was checked to see if it was set and if so could possibly
return AST_BRIDGE_COMPLETE.
(closes issue #15183)
Reported by: andrebarbosa
Tested by: andrebarbosa, tootai, loloski
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r195882 | mnicholson | 2009-05-21 10:33:55 -0500 (Thu, 21 May 2009) | 20 lines
Merged revisions 195881 via svnmerge from
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r195881 | mnicholson | 2009-05-21 10:25:50 -0500 (Thu, 21 May 2009) | 13 lines
This commit prevents cdr records with AST_CDR_FLAG_ANSLOCKED and AST_CDR_FLAG_LOCKED from being updated in certain cases.
This is accomplished by adding two functions to update the answer time and disposition of calls that checks for the proper lock flags. These functions are used in the ast_bridge_call() function so that ForkCDR(A) calls are respected.
This patch also modifies the way ast_bridge_call() chooses the cdr record to base the bridged_cdr on. Previously the first unlocked cdr record would be chosen, now instead the first cdr record is chosen and forked cdr records are moved to the bridge_cdr. This allows the original cdr record and any forked cdr records to be properly updated with answer and end times.
(closes issue #13797)
Reported by: sh0t
Tested by: sh0t
(closes issue #14744)
Reported by: deepesh
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r195370 | tilghman | 2009-05-18 15:52:33 -0500 (Mon, 18 May 2009) | 15 lines
Recorded merge of revisions 195366 via svnmerge from
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r195366 | tilghman | 2009-05-18 15:24:13 -0500 (Mon, 18 May 2009) | 8 lines
Add a similar dependency on SMDI for voicemail as already exists for ADSI.
(closes issue #14846)
Reported by: pj
Patches:
20090413__bug14846__1.4.diff.txt uploaded by tilghman (license 14)
20090507__issue14846__1.6.0.diff.txt uploaded by tilghman (license 14)
20090507__issue14846__1.6.1.diff.txt uploaded by tilghman (license 14)
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r192357 | kpfleming | 2009-05-05 15:18:21 +0200 (Tue, 05 May 2009) | 5 lines
Correct some flaws in the memory accounting code for stringfields and ao2 objects
Under some conditions, the memory allocation for stringfields and ao2 objects would not have supplied valid file/function names for MALLOC_DEBUG tracking, so this commit corrects that.
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r192318 | kpfleming | 2009-05-05 12:34:19 +0200 (Tue, 05 May 2009) | 5 lines
Properly account for memory allocated for channels and datastores
As in previous commits, when channels are allocated (with ast_channel_alloc) or datastores are allocated (with ast_datastore_alloc) properly account for the memory being owned by the caller, instead of the allocator function itself.
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r192279 | kpfleming | 2009-05-05 10:51:06 +0200 (Tue, 05 May 2009) | 5 lines
Ensure that string pools allocated to hold stringfields are properly accounted in MALLOC_DEBUG mode
This commit modifies the stringfield pool allocator to remember the 'owner' of the stringfield manager the pool is being allocated for, and ensures that pools allocated in the future when fields are populated are owned by that file/function.
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r191367 | tilghman | 2009-04-30 12:40:58 -0500 (Thu, 30 Apr 2009) | 3 lines
Detect eaccess (or euidaccess) before using it.
Reported by Andrew Lindh via the -dev list.
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r190725 | kpfleming | 2009-04-27 14:30:54 -0500 (Mon, 27 Apr 2009) | 13 lines
Merged revisions 190721 via svnmerge from
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r190721 | kpfleming | 2009-04-27 14:29:46 -0500 (Mon, 27 Apr 2009) | 7 lines
Fix 'inconsistent line endings' when autoconf 2.63 is used
Attempt to make configure script regeneration 'safe' using autoconf 2.63, which embeds a bare CR into the script, thus making Subversion complain about inconsistent line endings
This commit changes the MIME type of the configure script to be 'binary' thus making Subversion no longer inspect line endings, and as a bonus 'svn diff' will no longer try to generate diff output for it, which is not generally useful anyway.
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r190093 | tilghman | 2009-04-22 16:38:15 -0500 (Wed, 22 Apr 2009) | 14 lines
Merged revisions 190092 via svnmerge from
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r190092 | tilghman | 2009-04-22 16:35:03 -0500 (Wed, 22 Apr 2009) | 7 lines
Detect availability of pthread_rwlock_timedwrlock() before using it.
(closes issue #14930)
Reported by: tilghman
Patches:
20090420__bug14930.diff.txt uploaded by tilghman (license 14)
Tested by: mvanbaak, tilghman
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r190057 | jpeeler | 2009-04-22 16:15:55 -0500 (Wed, 22 Apr 2009) | 9 lines
Fix building of chan_h323 with gcc-3.3
There seems to be a bug with old versions of g++ that doesn't allow a structure
member to use the name list. Rename list member to group_list in ast_group_info
and change the few places it is used.
(closes issue #14790)
Reported by: stuarth
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r189629 | dbailey | 2009-04-21 09:28:04 -0500 (Tue, 21 Apr 2009) | 10 lines
Merged revisions 189601 via svnmerge from
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r189601 | dbailey | 2009-04-21 09:00:55 -0500 (Tue, 21 Apr 2009) | 3 lines
Add check in configure script to check for GLOB_NOMAGIC and GLOB_BRACE in glob.h
This allows config.c to compile when linked against uclibc that does not support these parameters
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r187483 | tilghman | 2009-04-09 13:40:01 -0500 (Thu, 09 Apr 2009) | 15 lines
Merged revisions 187428 via svnmerge from
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r187428 | tilghman | 2009-04-09 13:08:20 -0500 (Thu, 09 Apr 2009) | 8 lines
Race condition between ast_cli_command() and 'module unload' could cause a deadlock.
Add lock timeouts to avoid this potential deadlock.
(closes issue #14705)
Reported by: jamessan
Patches:
20090320__bug14705.diff.txt uploaded by tilghman (license 14)
Tested by: jamessan
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r187302 | tilghman | 2009-04-08 23:59:05 -0500 (Wed, 08 Apr 2009) | 14 lines
Merged revisions 187300-187301 via svnmerge from
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r187300 | tilghman | 2009-04-08 23:31:38 -0500 (Wed, 08 Apr 2009) | 3 lines
Add debugging mode for diagnosing file descriptor leaks.
(Related to issue #14625)
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r187301 | tilghman | 2009-04-08 23:32:40 -0500 (Wed, 08 Apr 2009) | 2 lines
Oops, missed this file in the last commit.
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r186321 | file | 2009-04-03 12:52:50 -0300 (Fri, 03 Apr 2009) | 12 lines
Merged revisions 186320 via svnmerge from
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r186320 | file | 2009-04-03 12:48:56 -0300 (Fri, 03 Apr 2009) | 5 lines
Fix a problem with the crypto variable definitions not actually being defined properly.
(closes issue #14804)
Reported by: jvandal
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r184147 | russell | 2009-03-24 20:42:10 -0500 (Tue, 24 Mar 2009) | 5 lines
Fix build issues on Mac OSX.
(closes issue #14714)
Reported by: ygor
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r183436 | dvossel | 2009-03-19 15:30:39 -0500 (Thu, 19 Mar 2009) | 13 lines
Merged revisions 183386 via svnmerge from
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r183386 | dvossel | 2009-03-19 14:40:07 -0500 (Thu, 19 Mar 2009) | 6 lines
Cleaning up a few things in detect disconnect patch
Initialized ast_call_feature in detect_disconnect to avoid accessing uninitialized memory. Cleaned up /param tags in features.h. No longer send dynamic features in ast_feature_detect.
issue #11583
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r183242 | russell | 2009-03-19 13:00:15 -0500 (Thu, 19 Mar 2009) | 10 lines
Merged revisions 183241 via svnmerge from
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r183241 | russell | 2009-03-19 12:52:52 -0500 (Thu, 19 Mar 2009) | 2 lines
Remove the use of RTLD_NOLOAD, as it is not behaving like expected.
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r183172 | dvossel | 2009-03-19 11:28:33 -0500 (Thu, 19 Mar 2009) | 20 lines
Merged revisions 183126 via svnmerge from
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r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009) | 17 lines
Allow disconnect feature before a call is bridged
feature.conf has a disconnect option. By default this option is set to '*', but it could be anything. If a user wishes to disconnect a call before the other side answers, only '*' will work, regardless if the disconnect option is set to something else. This is because features are unavailable until bridging takes place. The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different. This patch allows features to be detected from outside of the bridge, but not operated on. In this case, the disconnect feature can be detected before briding and handled outside of features.c.
(closes issue #11583)
Reported by: sobomax
Patches:
patch-apps__app_dial.c uploaded by sobomax (license 359)
11583.latest-patch uploaded by murf (license 17)
detect_disconnect.diff uploaded by dvossel (license 671)
Tested by: sobomax, dvossel
Review: http://reviewboard.digium.com/r/195/
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r182847 | russell | 2009-03-17 21:28:55 -0500 (Tue, 17 Mar 2009) | 52 lines
Merged revisions 182810 via svnmerge from
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r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines
Fix cases where the internal poll() was not being used when it needed to be.
We have seen a number of problems caused by poll() not working properly on
Mac OSX. If you search around, you'll find a number of references to using
select() instead of poll() to work around these issues. In Asterisk, we've
had poll.c which implements poll() using select() internally. However, we
were still getting reports of problems.
vadim investigated a bit and realized that at least on his system, even
though we were compiling in poll.o, the system poll() was still being used.
So, the primary purpose of this patch is to ensure that we're using the
internal poll() when we want it to be used.
The changes are:
1) Remove logic for when internal poll should be used from the Makefile.
Instead, put it in the configure script. The logic in the configure
script is the same as it was in the Makefile. Ideally, we would have
a functionality test for the problem, but that's not actually possible,
since we would have to be able to run an application on the _target_
system to test poll() behavior.
2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT
is not defined.
3) Change uses of poll() throughout the source tree to ast_poll(). I feel
that it is good practice to give the API call a new name when we are
changing its behavior and not using the system version directly in all cases.
So, normally, ast_poll() is just redefined to poll(). On systems where
AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll().
4) Change poll() in main/poll.c to be ast_internal_poll().
It's worth noting that any code that still uses poll() directly will work fine
(if they worked fine before). So, for example, out of tree modules that are
using poll() will not stop working or anything. However, for modules to work
properly on Mac OSX, ast_poll() needs to be used.
(closes issue #13404)
Reported by: agalbraith
Tested by: russell, vadim
http://reviewboard.digium.com/r/198/
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r182525 | kpfleming | 2009-03-17 09:38:11 -0500 (Tue, 17 Mar 2009) | 11 lines
Improve behavior of ast_answer() to not lose incoming frames
ast_answer(), when supplied a delay before returning to the caller, use ast_safe_sleep() to implement the delay. Unfortunately during this time any incoming frames are discarded, which is problematic for T.38 re-INVITES and other sorts of channel operations.
When a delay is not passed to ast_answer(), it still delays for up to 500 milliseconds, waiting for media to arrive. Again, though, it discards any control frames, or non-voice media frames.
This patch rectifies this situation, by storing all incoming frames during the delay period on a list, and then requeuing them onto the channel before returning to the caller.
http://reviewboard.digium.com/r/196/
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r181135 | jpeeler | 2009-03-10 23:06:44 -0500 (Tue, 10 Mar 2009) | 20 lines
Fix malloc debug macros to work properly with h323.
The main problem here was that cstdlib was undefining free thereby causing the
proper debug macros to not be used. ast_h323.cxx has been changed to call
ast_free instead to avoid the issue.
A few other issues were addressed:
- There were a few instances of functions improperly passing ast_free instead
of ast_free_ptr.
- Some clean up was done to avoid the debug macros intentionally being redefined.
(copied below from Kevin's commit, appreciate the help)
- disable astmm.h from doing anything when STANDALONE is defined, which is used
by the tools in the utils/ directory that use parts of Asterisk header files in
hackish ways; also ensure that utils/extconf.c and utils/conf2ael.c are
compiled with STANDALONE defined.
(closes issue #13593)
Reported by: pj
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r180373 | kpfleming | 2009-03-05 12:29:38 -0600 (Thu, 05 Mar 2009) | 15 lines
Merged revisions 180372 via svnmerge from
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r180372 | kpfleming | 2009-03-05 12:22:16 -0600 (Thu, 05 Mar 2009) | 9 lines
Fix problems when RTP packet frame size is changed
During some code analysis, I found that calling ast_rtp_codec_setpref() on an ast_rtp session does not work as expected; it does not adjust the smoother that may on the RTP session, in fact it summarily drops it, even if it has data in it, even if the current format's framing size has not changed. This is not good.
This patch changes this behavior, so that if the packetization size for the current format changes, any existing smoother is safely updated to use the new size, and if no smoother was present, one is created. A new API call for smoothers, ast_smoother_reconfigure(), was required to implement these changes.
Review: http://reviewboard.digium.com/r/184/
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r180032 | dvossel | 2009-03-03 17:21:18 -0600 (Tue, 03 Mar 2009) | 14 lines
app_read does not break from prompt loop with user terminated empty string
In app.c, ast_app_getdata is called to stream the prompts and receive DTMF input. If ast_app_getdata() receives an empty string caused by the user inputing the end of string character, in this case '#', it should break from the prompt loop and return to app_read, but instead it cycles through all the prompts. I've added a return value for this special case in ast_readstring() which uses an enum I've delcared in apps.h. This enum is now used as a return value for ast_app_getdata().
(closes issue #14279)
Reported by: Marquis
Patches:
fix_app_read.patch uploaded by Marquis (license 32)
read-ampersanmd.patch2 uploaded by dvossel (license 671)
Tested by: Marquis, dvossel
Review: http://reviewboard.digium.com/r/177/
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