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r378593 | jrose | 2013-01-04 17:14:54 -0600 (Fri, 04 Jan 2013) | 23 lines
res_srtp: Prevent a crash from occurring due to srtp_create failures in srtp_create
Under some circumstances, libsrtp's srtp_create function deallocates memory that
it wasn't initially responsible for allocating. Because we weren't initially
aware of this behavior, this memory was still used in spite of being unallocated
during the course of the srtp_unprotect function. A while back I made a patch
which would set this value to NULL, but that exposed a possible condition where
we would then try to check a member of the struct which would cause a segfault.
In order to address these problems, ast_srtp_unprotect will now set an error value
when it ends without a valid SRTP session which will result in the caller of
srtp_unprotect observing this error and hanging up the relevant channel instead of
trying to keep using the invalid session address.
(closes issue ASTERISK-20499)
Reported by: Tootai
Review: https://reviewboard.asterisk.org/r/2228/diff/#index_header
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r378565 | elguero | 2013-01-04 15:20:12 -0600 (Fri, 04 Jan 2013) | 27 lines
Fix SIP Notify Messages To Have The Proper IP Address In The FROM Field
On a multihomed server when sending a NOTIFY message, we were not figuring out
which network should be used to contact the peer.
This patch fixes the problem by calling ast_sip_ouraddrfor() and then
build_via() so that our NOTIFY message contains the correct IP address.
Also, a debug message is being added to help follow the call-id changes that
occur. This was helpful for confirming that the IP address was set properly
since the call-id contains the IP address. It also will be helpful for
troubleshooting purposes when following a call in the debug logs.
(closes issue ASTERISK-20805)
Reported by: Bryan Hunt
Tested by: Bryan Hunt, Michael L. Young
Patches:
asterisk-20805-notify-ip-v2.diff uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2255/
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r378585 | kmoore | 2013-01-04 16:19:16 -0600 (Fri, 04 Jan 2013) | 13 lines
Fix pjproject compilation in certain circumstances
On a fresh checkout of Asterisk 11, running make before ./configure
could cause the pjproject subdirectory to get in an odd state that
would prevent compilation. This patch by Tilghman prevents that from
occurring.
(closes issue ASTERISK-20681)
Patch-by: Tilghman Lesher
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r378557 | file | 2013-01-04 15:18:07 -0600 (Fri, 04 Jan 2013) | 11 lines
Don't pass STUN packets through the SRTP unprotect function.
(closes issue AST-1036)
Reported by: jbigelow
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r378543 | lathama | 2013-01-04 10:44:33 -0600 (Fri, 04 Jan 2013) | 6 lines
Doxygen Cleanups
Baseline clean up of formating to make room for extended documentation
(issue ASTERISK-20259)
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r378410 | mjordan | 2013-01-03 09:37:31 -0600 (Thu, 03 Jan 2013) | 13 lines
Prevent crashes in res_xmpp when receiving large messages
Similar to r378287, res_xmpp was marshaling data read from an external source
onto the stack. For a sufficiently large message, this could cause a stack
overflow. This patch modifies res_xmpp in a similar fashion to res_jabber by
removing the stack allocation, as it was unnecessary.
(issue ASTERISK-20658)
Reported by: wdoekes
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r378412 | file | 2013-01-03 09:40:21 -0600 (Thu, 03 Jan 2013) | 11 lines
Prevent exhaustion of system resources through exploitation of event cache
This patch changes res_xmpp to no longer cache events under certain circumstances.
(issue ASTERISK-20175)
Reported by: Russell Bryant, Leif Madsen, Joshua Colp
Tested by: kmoore
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r378414 | tilghman | 2013-01-03 10:04:11 -0600 (Thu, 03 Jan 2013) | 11 lines
Add aliases to the Directory.
This is an interesting feature that allows additional strings to be used to
search the Directory, primarily intended to be used with nicknames, but could
be used with affiliations and the like. Because the name field is used in
more than one place (such as email notifications), it is important that these
additional strings not be placed in the name field, but be specified
separately.
Review: https://reviewboard.asterisk.org/r/2244/
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r378322 | mjordan | 2013-01-02 12:11:59 -0600 (Wed, 02 Jan 2013) | 33 lines
Prevent exhaustion of system resources through exploitation of event cache
Asterisk maintains an internal cache for devices in the event subsystem. The
device state cache holds the state of each device known to Asterisk, such that
consumers of device state information can query for the last known state for
a particular device, even if it is not part of an active call. The concept of
a device in Asterisk can include entities that do not have a physical
representation. One way that this occurred was when anonymous calls are allowed
in Asterisk. A device was automatically created and stored in the cache for
each anonymous call that occurred; this was possible in the SIP and IAX2
channel drivers and through channel drivers that utilized the
res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif). These devices
are never removed from the system, allowing anonymous calls to potentially
exhaust a system's resources.
This patch changes the event cache subsystem and device state management to
no longer cache devices that are not associated with a physical entity.
(issue ASTERISK-20175)
Reported by: Russell Bryant, Leif Madsen, Joshua Colp
Tested by: kmoore
patches:
event-cachability-3.diff uploaded by jcolp (license 5000)
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r378288 | mjordan | 2013-01-02 09:39:42 -0600 (Wed, 02 Jan 2013) | 36 lines
Resolve crashes due to large stack allocations when using TCP
Asterisk had several places where messages received over various network
transports may be copied in a single stack allocation. In the case of TCP,
since multiple packets in a stream may be concatenated together, this can
lead to large allocations that overflow the stack.
This patch modifies those portions of Asterisk using TCP to either
favor heap allocations or use an upper bound to ensure that the stack will not
overflow:
* For SIP, the allocation now has an upper limit
* For HTTP, the allocation is now a heap allocation instead of a stack
allocation
* For XMPP (in res_jabber), the allocation has been eliminated since it was
unnecesary.
Note that the HTTP portion of this issue was independently found by Brandon
Edwards of Exodus Intelligence.
(issue ASTERISK-20658)
Reported by: wdoekes, Brandon Edwards
Tested by: mmichelson, wdoekes
patches:
ASTERISK-20658_res_jabber.c.patch uploaded by mmichelson (license 5049)
issueA20658_http_postvars_use_malloc2.patch uploaded by wdoekes (license 5674)
issueA20658_limit_sip_packet_size3.patch uploaded by wdoekes (license 5674)
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r378000 | seanbright | 2012-12-13 15:20:32 -0600 (Thu, 13 Dec 2012) | 8 lines
Make generate_exchange_uuid() always return the passed ast_str pointer.
I changed this code earlier to return NULL if it wasn't able to generate a UUID,
whereas the earlier code would always return the ast_str that was passed in.
Switch back to returning the ast_str, only set it to the empty string instead if
UUID generation fails. We still do a validity check later which will catch this
and blow up if necessary.
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r378001 | wedhorn | 2012-12-13 15:25:31 -0600 (Thu, 13 Dec 2012) | 9 lines
Minor fixes for chan_skinny
Whitespace, change SUBSTATE_ONHOOK to correct SKINNY_ONHOOK and
correct len of 2 strcmp in skinny_setdebug(). (see opticron's review
on https://reviewboard.asterisk.org/r/2240/)
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r378002 | rmudgett | 2012-12-13 15:28:15 -0600 (Thu, 13 Dec 2012) | 35 lines
confbridge: Fix MOH on simultaneous user entry to a new conference.
When two users entered a new conference simultaneously, one of the callers
hears MOH. This happened if two unmarked users entered simultaneously and
also if a waitmarked and a marked user entered simultaneously.
* Created a confbridge internal MOH API to eliminate the inlined MOH
handling code. Note that the conference mixing bridge needs to be locked
when actually starting/stopping MOH because there is a small window
between the conference join unsuspend MOH and actually joining the mixing
bridge.
* Created the concept of suspended MOH so it can be interrupted while
conference join announcements to the user and DTMF features can operate.
* Suspend any MOH until the user is about to actually join the mixing
bridge of the conference. This way any pre-join file playback does not
need to worry about MOH.
* Made post-join actions only play deferred entry announcement files.
Changing the user/conference state during that time is not protected or
controlled by the state machine.
(closes issue ASTERISK-20606)
Reported by: Eugenia Belova
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/2232/
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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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r377035 | oej | 2012-12-03 10:45:49 -0600 (Mon, 03 Dec 2012) | 2 lines
Formatting fixes
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r377040 | rmudgett | 2012-12-03 11:10:40 -0600 (Mon, 03 Dec 2012) | 16 lines
Fix CCSS CLI commands and logger level not unregistered.
(issue ASTERISK-20649)
Reported by: Corey Farrell
Patches:
ccss-cleanup-all.patch (license #5909) patch uploaded by Corey Farrell
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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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r376092 | mmichelson | 2012-11-08 16:10:29 -0600 (Thu, 08 Nov 2012) | 18 lines
Fix a "set but not used" warning on newer gccs.
Turns out the "helpful" setting of ms and res in this
macro is completely useless after the timeout antipattern
fix.
If you're a new guy looking to write code, don't write
a macro like this one.
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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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Its perfectly acceptable to have a gateway session unreserved when we go to
first allocate one. Unreffing the reserved gateway session - when its NULL -
will result in an assertion error.
This problem was caught by the Asterisk Test Suite (once we had enough of the
debugging flags enabled)
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could not be registered.
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)
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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)
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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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This patch resolves a number of ref leaks that occur primarily on Asterisk
shutdown. It adds a variety of shutdown routines to core portions of
Asterisk such that they can reclaim resources allocate duringd initialization.
Review: https://reviewboard.asterisk.org/r/2137
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Start adding configuration file linking and pages. Add module loading doxygen block.
Breaking up commits to keep it easy to track
(issue ASTERISK-20259)
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from dialplan.
(closes issue ASTERISK-17136)
Reported by: kenner
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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.
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The opinion of development was that it is both improper to have Matt's
personal email address used in the source and that the command wouldn't
be useful without it.
(closes issue AST-467)
Reported by: Malcolm Davenport
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in res_xmpp on unload.
This patch fixes an issue where hangup flags were not being reset on a
channel, affecting subsequent use of that channel. The patch also adds some
additional cleanup to res_xmpp to fix an issue with reloading the module.
(closes ASTERISK-20360)
Reported by: Noah Engelberth
Tested by: beagles
Review: https://reviewboard.asterisk.org/r/2134/
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musiconhold.
(issue ASTERISK-17367)
Reported by: oej
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chan_sip.
(closes issue ASTERISK-20439)
Reported by: sruffell
Patches:
0001-chan_sip-websocket-support-is-an-optional-API.patch uploaded by sruffell (license 5417)
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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/
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determine it failed when in reality it did not.
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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improper SDP being produced.
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)
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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)
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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.
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(closes issue ASTERISK-20361)
Reported by: Noah Engelberth
Review: https://reviewboard.asterisk.org/r/2108/
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chan_gtalk, chan_jingle, and res_jabber are now deprecated in favor of
using chan_motif and res_xmpp. They are a feature-equivalent
replacement and are written to be more easily maintainable.
(closes issue ASTERISK-20298)
Review: https://reviewboard.asterisk.org/r/2082/
Reported-by: Leif Madsen
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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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r372327 | rmudgett | 2012-09-05 12:33:11 -0500 (Wed, 05 Sep 2012) | 15 lines
Fix RTP/RTCP read error message confusion.
The RTP/RTCP read error message can report "fail: success" when the
read failure is because of an ICE failure.
* Changed __rtp_recvfrom() to generate a PJ ICE message when ICE fails.
* Changed RTP/RTCP read error message to indicate an unspecified error
when errno is zero.
(closes issue ASTERISK-20288)
Reported by: Joern Krebs
Patches:
jira_asterisk_20288_err_msg.patch (license #5621) patch uploaded by rmudgett (modified)
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r372328 | rmudgett | 2012-09-05 12:35:20 -0500 (Wed, 05 Sep 2012) | 1 line
Fix coding guidelines issue with a recent commit.
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