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r377658 | kmoore | 2012-12-10 10:56:37 -0600 (Mon, 10 Dec 2012) | 20 lines
Ensure ReceiveFax provides a CED tone via T.38
When using res_fax_digium, the T.38 CED tone was not being provided
properly which would cause some incoming faxes to fail. This was not an
issue with res_fax_spandsp since it does not strictly honor the
send_ced flag and sends the CED tone whenever receiving a T.38 fax.
(closes issue FAX-343)
Reported-by: Benjamin Tietz
Patch-by: Kinsey Moore
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r377260 | file | 2012-12-05 10:51:58 -0600 (Wed, 05 Dec 2012) | 25 lines
Fix a SIP request memory leak with TLS connections.
During the TLS re-work in chan_sip some TLS specific code was moved
into a separate function. This function operates on a copy of the
incoming SIP request. This copy was never deinitialized causing a
memory leak for each request processed.
This function is now given a SIP request structure which it can use
to copy the incoming request into. This reduces the amount of memory
allocations done since the internal allocated components are reused
between packets and also ensures the SIP request structure is
deinitialized when the TLS connection is torn down.
(closes issue ASTERISK-20763)
Reported by: deti
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r377263 | jrose | 2012-12-05 11:17:06 -0600 (Wed, 05 Dec 2012) | 21 lines
res_srtp: Fix a crash caused by srtp_dealloc on an already dealloced session
When srtp_create fails, the session may be dealloced or just not alloced. At
the same time though, the session pointer might not be set to NULL in this
process and attempting to srtp_dealloc it again will cause a segfault. This
patch checks for failure of srtp_create and sets the session pointer to NULL
if it fails.
(closes issue ASTERISK-20499)
Reported by: tootai
Review: https://reviewboard.asterisk.org/r/2228/
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r376998 | oej | 2012-12-03 03:35:55 -0600 (Mon, 03 Dec 2012) | 4 lines
Formatting changes
Found a large amount of missing {} in the code before patching in another branch
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r376562 | dlee | 2012-11-20 16:06:05 -0600 (Tue, 20 Nov 2012) | 8 lines
Added missing newlines to websocket ast_logs.
Without these newlines, log messages just continue tacking onto the same
line, and do not flush immediately.
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r376131 | file | 2012-11-11 11:15:47 -0600 (Sun, 11 Nov 2012) | 16 lines
Remove a fixed size limitation for producing SDP and change how ICE support is disabled by default.
With ICE support enabled in chan_sip and a large number of interfaces on the system it was
possible for the produced SDP to be truncated due to some fixed size buffers. These buffers
have now been changed so they will dynamically grow as needed.
ICE support is now also enabled by default in res_rtp_asterisk to provide a smoother experience
for chan_motif users where it is required. To maintain the previous behavior in chan_sip it is
no longer enabled by default there.
(closes issue ASTERISK-20643)
Reported by: coopvr
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r375993 | mmichelson | 2012-11-07 11:01:13 -0600 (Wed, 07 Nov 2012) | 30 lines
Fix misuses of timeouts throughout the code.
Prior to this change, a common method for determining if a timeout
was reached was to call a function such as ast_waitfor_n() and inspect
the out parameter that told how many milliseconds were left, then use
that as the input to ast_waitfor_n() on the next go-around.
The problem with this is that in some cases, submillisecond timeouts
can occur, resulting in the out parameter not decreasing any. When this
happens thousands of times, the result is that the timeout takes much
longer than intended to be reached. As an example, I had a situation where
a 3 second timeout took multiple days to finally end since most wakeups
from ast_waitfor_n() were under a millisecond.
This patch seeks to fix this pattern throughout the code. Now we log the
time when an operation began and find the difference in wall clock time
between now and when the event started. This means that sub-millisecond timeouts
now cannot play havoc when trying to determine if something has timed out.
Part of this fix also includes changing the function ast_waitfor() so that it
is possible for it to return less than zero when a negative timeout is given
to it. This makes it actually possible to detect errors in ast_waitfor() when
there is no timeout.
(closes issue ASTERISK-20414)
reported by David M. Lee
Review: https://reviewboard.asterisk.org/r/2135/
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r375994 | mmichelson | 2012-11-07 11:08:44 -0600 (Wed, 07 Nov 2012) | 3 lines
Remove some debugging that accidentally made it in the last commit.
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Currently, if an acknowledgement of a timer fails Asterisk will not realize
that a serious error occurred and will continue attempting to use the timer's
file descriptor. This can lead to situations where errors stream to the
CLI/log file. This consumes significant resources, masks the actual problem
that occurred (whatever caused the timer to fail in the first place), and
can leave channels in odd states.
This patch propagates the errors in the timing resource modules up through
the timer core, and makes users of these timers handle acknowledgement
failures. It also adds some defensive coding around the use of timers
to prevent using bad file descriptors in off nominal code paths.
Note that the patch created by the issue reporter was modified slightly for
this commit and backported to 1.8, as it was originally written for
Asterisk 10.
Review: https://reviewboard.asterisk.org/r/2178/
(issue ASTERISK-20032)
Reported by: Jeremiah Gowdy
patches:
jgowdy-timerfd-6-22-2012.diff uploaded by Jeremiah Gowdy (license 6358)
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On some systems the optional API support uses the GCC compiler attribute "weakref" to provide its
functionality. This code changes the function names and prefixes "__" to the front. The
res_http_websocket exports file did not take this into account, thereby not allowing those functions
to be global and ultimately found.
(closes issue ASTERISK-20631)
Reported by: danjenkins
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Unlike all other calendar modules, res_calendar_ews fails to extract the Body
information for a calendar item. This is due, in part, to a quirk in the
schema in the XML - not only does a CalendarItem contain a Body element, but
the CalendarItem exists as a descendant of a different Body element. The neon
parser was erroneously skipping all Body elements.
This patch fixes that by bypassing Body elements that are not a child of
CalendarItem, and parsing the Body element out if it is a child.
Note that the original patch by Terry Wilson only needed slight modifications
to make it properly pull the Body information out; as such, while I've linked
to the patch that I uploaded for Dmitry, I've attributed the patch to Terry.
(closes issue ASTERISK-19738)
Reported by: Dmitry Burilov
Tested by: Dmitry Burilov
patches:
calendar_ews_body_2012_10_29.diff uploaded by Terry Wilson (license 6283)
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Remove the "RTP Read too short" warning for RTP keepalives. Remove the
the warning about the application delimiter switch from pipe to comma.
(You should've done this by now.) Make cdr_odbc report more when an
insert fails. Make chan_sip warn less when the peer wants SRTP (and we
don't) or sends a zero port to disable a media type.
Review: https://reviewboard.asterisk.org/r/2167
(closes issue ASTERISK-20538)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375137 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Update and extend the configuration_file group and enable linking to the resource. Update title that was left behind many years ago.
(issue ASTERISK-20259)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375003 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Since there are a number of legacy devices out there that fail to handle ICE
candidates properly (which is a nice way of saying something much uglier),
disable it by default.
Support for ICE candidates can be enabled in rtp.conf using the icesupport
setting.
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pjproject, in order to solve build problems on Windows [1], undefines s_addr in
one of it's headers that is included in res_rtp_asterisk.c. On Solaris s_addr
is not a structure member, but defined to map to the real strucuture member,
therefore when building on Solaris it's possible to get build errors like:
[CC] res_rtp_asterisk.c -> res_rtp_asterisk.o
In file included from /export/home/admin/asterisk-11-svn/include/asterisk/stun.h:29,
from res_rtp_asterisk.c:51:
/export/home/admin/asterisk-11-svn/include/asterisk/network.h: In function `inaddrcmp':
/export/home/admin/asterisk-11-svn/include/asterisk/network.h:92: error: structure has no member named `s_addr'
/export/home/admin/asterisk-11-svn/include/asterisk/network.h:92: error: structure has no member named `s_addr'
res_rtp_asterisk.c: In function `ast_rtp_on_ice_tx_pkt':
res_rtp_asterisk.c:706: warning: dereferencing type-punned pointer will break strict-aliasing rules
res_rtp_asterisk.c:710: warning: dereferencing type-punned pointer will break strict-aliasing rules
res_rtp_asterisk.c: In function `rtp_add_candidates_to_ice':
res_rtp_asterisk.c:1085: error: structure has no member named `s_addr'
make[2]: *** [res_rtp_asterisk.o] Error 1
make[1]: *** [res] Error 2
make[1]: Leaving directory `/export/home/admin/asterisk-11-svn'
gmake: *** [_cleantest_all] Error 2
Unfortunately, in order to make this work, I also had to make sure pjproject
only used the typdef pj_in_addr and not the struct pj_in_addr so that when
building Asterisk I could "typedef struct in_addr pj_in_addr". It's possible
then that the library and users of those interfaces in Asterisk have a different
idea about the type of the argument, while on the surface it looks like they are
all 32 bit big endian values.
[1] http://trac.pjsip.org/repos/changeset/484
(issues ASTERISK-20366)
Reported by: Ben Klang
Tested by: Ben Klang, mjordan
patches:
0001-pjproject-Fix-for-Solaris-builds.-Do-not-undef-s.patch uploaded by Shaun Ruffell (license 5417)
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While XEP-0115 states that the node and ver attributes are both required, some
devices fail to provide either field. Prior to this patch, failure to provide
the node or ver attribute would cause a crash in res_xmpp. While failing to
provide the node or ver attribute is technically invalid, since this
information is not utilized by Asterisk except for reporting purposes, for
interoperability reasons, we continue to process the capability stanza anyways.
(closes issue ASTERISK-20495)
Reported by: Martin W
Tested by: Martin W
patches:
20495.patch uploaded by Martin W (license #6434)
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When using the channel technology agnostic application/AMI command MessageSend,
the "From" field is technically optional for the SIP channel driver. However,
if being sent by the XMPP resource module (either res_xmpp or res_jabber), the
"From" field is necessary, and must correspond to a defined account. This
patch updates the documentation for this application/AMI command to reflect
this.
(closes issue ASTERISK-20405)
Reported by: Leif Madsen
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The AMI DBDelTree command will return Success/Key tree deleted successfully even
if the given key does not exist. The CLI command 'database deltree' had a
similar problem, but was saved because it actually responded with '0 database
entries removed'. AGI had a slightly different error, where it would return
success if the database was unavailable.
This came from confusion about the ast_db_deltree retval, which is -1 in the
event of a database error, or number of entries deleted (including 0 for
deleting nothing).
* Changed some poorly named res variables to num_deleted
* Specified specific errors when calling ast_db_deltree (database unavailable
vs. entry not found vs. success)
* Fixed similar bug in AGI database deltree, where 'Database unavailable'
results in successful result
(closes issue AST-967)
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/2138/
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The res_jabber resource module uses the ASTOBJ library for managing its ref
counted objects. After calling ASTOBJ_CONTAINER_FIND to locate a buddy object,
the pointer to the object has to be checked to see if the buddy existed.
Prior to this patch, the buddy object was not checked for NULL; with this patch
in both aji_client_info_handler and aji_dinfo_handler the pointer is checked
before used and, if no buddy object was found, the handlers return an error
code.
This patch does not take the approach that our JID can be used to log in from
another resource. If that approach is desired, an improvement could be made to
this patch to create the buddy on the fly. This patch seeks only to prevent
Asterisk from crashing.
FYI: In Asterisk 11+, you really should be using res_xmpp. It does not have
this problem, as it moved to the astobj2 library.
Note that multiple people have proposed patches for this issue; the patch being
committed here is based on those.
(closes issue ASTERISK-19532)
Reported by: Karsten Wemheuer
Tested by: Byron Clark
patches:
fix-jabber uploaded by Karsten Wemheuer (license #5930)
xmpp_no_crash_with_ejabberd.patch uploaded by Byron Clark (license #6157)
(closes issue ASTERISK-19557)
Reported by: ulugutz
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* Added AMI event documentation for AsyncAGI and AGIExec events.
(closes issue ASTERISK-20318)
Reported by: Dan Cropp
Patches:
res_agi_patch.txt (license #6422) patch uploaded by Dan Cropp
modified for trunk.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374075 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch changes get_agi_cmd so that the return can be checked
to differentiate between an empty list success and something that
triggered an error. This in turn allows launch_asyncagi to detect
these errors and break free from the command processing loop so
that the async agi can be ended more cleanly
(closes issue ASTERISK-20109)
Reported by: Jeremiah Gowdy
Patches: jgowdy-7-9-2012.diff uploaded by Jeremiah Gowdy (license 6358)
(Modified by me to fix some logical issues and apply to trunk)
Review: https://reviewboard.asterisk.org/r/2117/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373608 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When sending RTP packets via multicast the amount of data sent is stored in a variable and returned
from the write function. This is incorrect as any non-zero value returned is considered a failure while
a return value of 0 is success. For callers (such as ast_streamfile) that checked the return value
they would have considered it a failure when in reality nothing went wrong and it was actually a success.
The write function for the multicast RTP engine now returns -1 on failure and 0 on success, as it should.
(closes issue ASTERISK-17254)
Reported by: wybecom
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The H.264 format attribute module compares two format attribute structures to determine if they are
compatible or not. In some instances it was possible for this check to determine that both structures
were incompatible when they actually should be considered compatible. This check has now been made even
more permissive by assuming that if no attribute information is available the two structures are compatible.
If both structures contain attribute information a base level comparison of the H.264 IDC value is done to
see if they are compatible or not.
The above issue uncovered a secondary issue in chan_sip where the SDP being produced would be incorrect if
the formats were considered incompatible. This has now been fixed by checking that all information required
to produce the SDP is available instead of assuming it is.
(closes issue ASTERISK-20464)
Reported by: Leif Madsen
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This patch removes the turnport configuration property and changes the
turnaddr property to be a combined host[:port] configuration string. The
patch also modifies the documentation in the example configuration to
reflect the property changes and adds some additional text indicating how
the STUN port is configured.
(closes issue ASTERISK-20344)
Reported by: beagles
Tested by: beagles
Review: https://reviewboard.asterisk.org/r/2111/
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* Whitespace, doc-blocks, spelling, case, missing and incorrect tags.
* Add cleanup to Makefile for the Doxygen configuration update
* Start updating Doxygen configuration for cleaner output
* Enable inclusion of configuration files into documentation
* remove mantisworkflow...
* update documentation README
* Add markup to Tilghman's email and talk with him about updating his email, he knows...
* no code changes on this commit other than the mentioned Makefile change
(issue ASTERISK-20259)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Doxygen updates including mistakes, misspellings, missing parameters, updates for Doxygen style. Some missing txt file links are removed but their content or essense will be included in some later updates. A majority of the txt files were removed in the 1.6 era but never noted. The HR and EXTREF are simple changes that make the documentation more compatable with more versions of Doxygen.
Further updates coming.
(issue ASTERISK-20259)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373330 65c4cc65-6c06-0410-ace0-fbb531ad65f3
As mentioned on the review for this, WebRTC has moved towards choosing
DTLS-SRTP as the mechanism for key exchange for SRTP. This commit adds
support for this but makes it available for normal SIP clients as well.
Testing has been done to ensure that this introduces no regressions with
existing behavior and also that it functions as expected.
Review: https://reviewboard.asterisk.org/r/2113/
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Before this, attempting to unload res_curl.so would warn you about the first
module it found that was dependent. We now warn about all of the loaded modules
instead.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373203 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Removes "res_rtp_asterisk.c:706: warning: dereferencing type-punned pointer
will break strict-aliasing rules" warning from the build on 32-bit platforms.
The problem is that 'size' was referenced aliased to both (pj_size_t *) and
(pj_ssize_t *). Now just make a copy of size that is the right type so there
isn't any pointer aliasing happening.
It also adds comments and asserts regarding what looks like an inappropriate
use of pj_sock_sendto, but is actually totally fine.
(closes issue ASTERISK-20368)
Reported by: Shaun Ruffell
Tested by: Michael L. Young
Patches:
0001-res_rtp_asterisk-Eliminate-type-punned-pointer-build.patch uploaded by Shaun Ruffell (license 5417)
slightly modified by David M. Lee.
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Fixes a build regression introduced in r369517 "Add support for ICE/STUN/TURN
in res_rtp_asterisk and chan_sip." [1].
[1] http://svnview.digium.com/svn/asterisk?view=revision&revision=369517
When compiling asterisk in parallel like:
$ make -j 10
It's possible to get errors like the following:
.pjlib-util-test-x86_64-unknown-linux-gnu.depend:120: *** missing separator. Stop.
make[4]: *** [depend] Error 2
make[3]: *** [dep] Error 1
make[2]: *** [/home/sruffell/asterisk-working/res/pjproject/pjnath/lib/libpjnath-x86_64-unknown-linux-gnu.a] Error 2
make[3]: warning: jobserver unavailable: using -j1. Add `+' to parent make rule.
This is because the build system is trying to build each of the libraries in
pjproject in parallel. Now the build will build pjproject in a single job and
link the results into res_asterisk_rtp.
Parallel builds, on one test system, saves ~1.5 minutes from a default Asterisk
build:
Single job:
$ git clean -fdx >/dev/null && time ( ./configure >/dev/null 2>&1 && make >/dev/null 2>&1 )
real 2m34.529s
user 1m41.810s
sys 0m15.970s
Parallel make:
$ git clean -fdx >/dev/null && time ( ./configure >/dev/null 2>&1 && make -j10 >/dev/null 2>&1 )
real 1m2.353s
user 2m39.120s
sys 0m18.850s
(closes issue ASTERISK-20362)
Reported by: Shaun Ruffel
Patches:
0001-res_asterisk_rtp-Fix-build-error-when-using-parallel.patch uploaded by Shaun Ruffel (License #5417)
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Merged revisions 372609 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372610 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The previous fix still would look in the static_RTP_PT table, which
is inappropriate since we specifically want to find a codec that has
been negotiated.
(closes issue ASTERISK-20296)
reported by NITESH BANSAL
Patches:
codec_negotiation.patch Uploaded by NITESH BANSAL (License #6418)
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Merged revisions 372311 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372319 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In Asterisk 1.4+, a fix was put in place to increment the sequence number for
retransmitted DTMF end packets. With the introduction of the RTP engine API in
1.8, the sequence number was no longer being incremented. This patch fixes this
regression as well as cleans up a few lines that were not doing anything.
(closes issue ASTERISK-20295)
Reported by: Nitesh Bansal
Tested by: Michael L. Young
Patches:
01_rtp_event_seq_num.patch uploaded by Nitesh Bansal (license 6418)
asterisk-20295-dtmf-fix-cleanup.diff uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2083/
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Merged revisions 372185 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 372198 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 372199 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A change for Asterisk 11 caused a check for failure to incorrectly check the return
value. This resulted in the possibility of transmitting media that a party had not
negotiated. If this media happened to be G.729, then this could potentially result
in one-way audio if no G.729 translators are installed.
(closes issue ASTERISK-20296)
reported by NITESH BANSAL
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Merged revisions 372118 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372119 65c4cc65-6c06-0410-ace0-fbb531ad65f3
pj_thread_register() takes a parameter of type pj_thread_desc.
It was assumed that pj_thread_register either used this item
temporarily or made a copy of it. Unfortunately, all it does is
keep a pointer to the structure in thread-local storage. This
means that if our pj_thread_desc goes out of scope, then pjlib
will be referencing bogus data quite often, most commonly on
operations involving a pj_mutex_t.
In our case, our pj_thread_desc was on the stack and went out
of scope very shortly after registering our thread with pjlib.
With this change, the pj_thread_desc is stored in thread-local
storage so the pointer that pjlib keeps in thread-local storage
will reference legitimate memory.
(closes issue ASTERISK-20237)
reported by Jeremy Pepper
Patches:
ASTERISK-20237.patch uploaded by Mark Michelson (license #5049)
Tested by Jeremy Pepper
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Merged revisions 371571 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371572 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Some core support modules and compiler options were no longer tagged with a
module support level. This patch adds 'core' back to those options.
Note that this patch modifies a few of the patches provided by Andrew Latham
slightly. res_curl and res_fax are both 'core' supported modules.
(closes issue ASTERISK-20215)
Reported by: Andrew Latham
Tested by: mjordan
Patches:
astcanary.diff (license #5985) uploaded by Andrew Latham
cflagsxml.diff (license #5985) uploaded by Andrew Latham
curl_fax.diff (license #5985) uploaded by Andrew Latham
soundsxml.diff (license #5985) uploaded by Andrew Latham
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Merged revisions 371507 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
While building up a new install to test chan_motif, I ran into a failure
due to icesupport being disabled. This was due to me not having an
rtp.conf. It was intended in the code for it to be enabled by default,
but it was only applied if rtp.conf existed.
This patch updates res_rtp_asterisk to be consistent in how it handles
defaults. A few options didn't have their default values set globally,
including icesupport. They are now set and icesupport is enabled by
default, even if you do not have an rtp.conf.
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Merged revisions 371425 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371428 65c4cc65-6c06-0410-ace0-fbb531ad65f3
