Terry Wilson [Mon, 17 Oct 2011 17:35:23 +0000 (17:35 +0000)]
Initialize variables before calling parse_uri
If parse_uri was called with an empty URI, some pointers would be
modified and an invalid read could result. This patch avoids calling
parse_uri with an empty contact uri when parsing REGISTER requests.
Kevin P. Fleming [Fri, 14 Oct 2011 21:36:06 +0000 (21:36 +0000)]
Change the internal name of the menuselect options that are used to control
whether modules are embedded or not; using just the bare category name led to
accidentally enabling these options when users used the wrong "--enable"
operation on the menuselect command line.
Now the internal option names are prefixed with "EMBED_", so they won't be
the same as the name of the category containing the modules they control
the embedding of.
Kinsey Moore [Fri, 14 Oct 2011 20:49:39 +0000 (20:49 +0000)]
Quiet RTCP Receiver Reports during fax transmission
RTCP is now disabled for "inactive" RTP audio streams during SIP T.38 sessions.
The ability to disable RTCP streams in res_rtp_asterisk was missing, so this
code was added to support the bug fix.
Terry Wilson [Thu, 13 Oct 2011 00:05:17 +0000 (00:05 +0000)]
Don't skip the query field on a realtime multi query
There is no documented reason to not add the query field to the varlist
returned by a realtime multi query, despite the config category being
set to its value. Of course, there is no documentation that the category
should be set to the value either. There is lots of no documentation
when it comes to realtime. But, other engines do not skip this field so
I am forcing this backend to follow the convention, because not doing so
is very silly.
Terry Wilson [Wed, 12 Oct 2011 20:19:36 +0000 (20:19 +0000)]
Update SIP realtime fullcontact regardless of caching
We should update the fullcontact field in the realtime table whether or
not rtcachefriends is set. There is no reason to treat a non-cached
realtime entity differently than a cached in this regard.
Richard Mudgett [Wed, 12 Oct 2011 17:49:19 +0000 (17:49 +0000)]
Update MeetMe p and X option documentation when interacting with the s option.
ASTERISK-12175 changed the p and X options to not interfere with the s
option when they are used together. It makes more sense for the s option
to have priority for the DTMF '*' key since it cannot change its
activation code. Otherwise, you could not use option s with the p or X
options.
Richard Mudgett [Tue, 11 Oct 2011 18:23:14 +0000 (18:23 +0000)]
Convert registered AMI actions to ao2 objects.
* Fixed race between calling an AMI action callback and unregistering that
action. Refixes ASTERISK-13784 broken by ASTERISK-17785 change.
* Fixed potential memory leak if an AMI action failed to get registered
because is already was registered. Part of the ao2 conversion.
* Fixed AMI ListCommands action not walking the actions list with a lock
held.
* Fix usage of ast_strdupa() and alloca() in loops. Excess stack usage.
* Fix AMI Originate action Variable header requiring a space after the
header colon. Reported by Yaroslav Panych on the asterisk-dev list.
* Increased the number of listed variables allowed per AMI Originate
action Variable header to 64.
* Fixed AMI GetConfigJSON action output format.
* Fixed usage of res contents outside of scope in append_channel_vars().
* Fixed inconsistency of config file channelvars option. The values no
longer accumulate with every channelvars option in the config file. Only
the last value is kept to be consistent with the CLI "manager show
settings" command.
Tzafrir Cohen [Tue, 11 Oct 2011 00:43:14 +0000 (00:43 +0000)]
Update SHA1 code to RFC 6234
RFC 6234 is an update to RFC 3174 from which the code was originally taken.
It has a slightly better code, and a better phrased license (simple 3-clause
BSD).
* main/sha1.c is sha1.c from RFC 6234 with formatting changes only.
* include/asterisk/sha1.h merges sha.h and sha-private.h from RFC 6234.
* Removed unused include of asterisk/sha1.h from main/channels.c
Matthew Jordan [Mon, 10 Oct 2011 20:23:48 +0000 (20:23 +0000)]
Updated chan_sip to place calls on hold if SDP address in INVITE is ANY
This patch fixes the case where an INVITE is received with c=0.0.0.0 or ::.
In this case, the call should be placed on hold. Previously, we checked for
the address being null; this patch keeps that behavior but also checks for
the ANY IP addresses.
Review: https://reviewboard.asterisk.org/r/1504/
(closes issue ASTERISK-18086)
Reported by: James Bottomley
Tested by: Matt Jordan
Load the proper XML documentation when multiple modules document the same application.
This patch adds an optional "module" attribute to the XML documentation spec
that allows the documentation processor to match apps with identical names from
different modules to their documentation. This patch also fixes a number of
bugs with the documentation processor and should make it a little more
efficient. Support for multiple languages has also been properly implemented.
Richard Mudgett [Thu, 6 Oct 2011 22:47:50 +0000 (22:47 +0000)]
Fix regression in configure script for libpri capability checks.
JIRA AST-598 added the PRI_L2_PERSISTENCE option to fix BRI PTMP TE layer
2 persistence issues with some telcos. ASTERISK-18535 attempted to fix
the unexpected requirement that libpri *must* have that feature to work
with Asterisk. The AST_EXT_LIB_SETUP_DEPENDENT lines made the PRI
optional features required. Unfortunately, I thought
AST_EXT_LIB_SETUP_DEPENDENT didn't do anything useful for libpri and
deleted those lines for libpri. The result was the HAVE_PRI_xxx defines
that control the ability to use optional libpri features were also
deleted.
* Created AST_EXT_LIB_SETUP_OPTIONAL configuration macro to allow optional
features in a library that the source code could take advantage of if the
code supports the feature.
Richard Mudgett [Thu, 6 Oct 2011 17:49:38 +0000 (17:49 +0000)]
Fix debugging messages generated by 'udptl debug'.
* Makes chan_sip set the tag to the channel name.
* Fixes received debug message sequence number.
* Removed tx/rx debug message type since it was hard coded to 0.
* Made udptl.c logged message header consistent if possible: "UDPTL (%s): ".
* Removed unused rx_expected_seq_no from struct ast_udptl.
(closes issue ASTERISK-18401)
Reported by: Kevin P. Fleming
Patches:
jira_asterisk_18401_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: Matthew Nicholson
Leif Madsen [Wed, 5 Oct 2011 21:30:11 +0000 (21:30 +0000)]
Update prep_tarball script to download pre-exported documentation.
I've updated the prep_tarball script to now download the pre-exported documentation
from the Asterisk wiki. This will give us more control over what is being included
in the tarball releases, and will make both the PDF and HTML exported documentation
look much better (especially when viewing from a console).
Jonathan Rose [Tue, 4 Oct 2011 19:33:12 +0000 (19:33 +0000)]
Removes improper use of sound 'and' in German language mode from application saynumber
Asterisk would say 'Five hundert und sechs und zwanzig' instead of 'Five hundert sechs
und zwanzig'... which is both weird sounding and wrong. This patch makes sure Asterisk
will only say the 'and' word between the single digit and double digit places.
Jonathan Rose [Tue, 4 Oct 2011 14:01:05 +0000 (14:01 +0000)]
Reverting revision 333265 due to component connection problems it introduces.
I'm going to attempt some generic res_jabber cleanup and come up with a new fix for this
problem, but first it seems prudent to remove this rather broad attempt to fix it and
instead approach this problem either from the same angle but looking only at canceling
(or possibly rescheduling) the send when we absolutely know it will cause a segfault
or, if that can't be easily accomplished, strictly from the devstate side of things.
Also, I'm pretty sure a lot of the code in res_jabber isn't thread safe.
Terry Wilson [Mon, 3 Oct 2011 18:40:52 +0000 (18:40 +0000)]
Properly ignore AST_CONTROL_UPDATE_RTP_PEER in more places
After the change in r336294, the new AST_CONTROL_UPDATE_RTP_PEER frame
is sent when a re-invite happens. If we receive a re-invite from a device
the waitstream_core was not aware of the new control frame and would drop
the call.
Richard Mudgett [Fri, 30 Sep 2011 22:05:10 +0000 (22:05 +0000)]
Fix segfault in analog_ss_thread() not checking ast_read() for NULL.
NOTE: The problem was reported against v1.6.2. It is unlikely to ever
happen on v1.8 and above since chan_dahdi.c:analog_ss_thread() is unlikely
to be used. The version in sig_analog.c has largely replaced it.
(closes issue ASTERISK-18648)
Reported by: Stephan Bosch
Patches:
jira_asterisk_18648_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: Stephan Bosch
The SIP_HEADER function only works on the the initial SIP INVITE. The documentation was updated
in trunk, but not in 1.8 or 10, so I'm making them match.
Richard Mudgett [Mon, 26 Sep 2011 19:30:39 +0000 (19:30 +0000)]
Fix deadlock when using dummy channels.
Dummy channels created by ast_dummy_channel_alloc() should be destoyed by
ast_channel_unref(). Using ast_channel_release() needlessly grabs the
channel container lock and can cause a deadlock as a result.
* Analyzed use of ast_dummy_channel_alloc() and made use
ast_channel_unref() when done with the dummy channel. (Primary reason for
the reported deadlock.)
* Made app_dial.c:dial_exec_full() not call ast_call() holding any channel
locks. Chan_local could not perform deadlock avoidance correctly.
(Potential deadlock exposed by this issue. Secondary reason for the
reported deadlock since the held lock was part of the deadlock chain.)
* Fixed some uses of ast_dummy_channel_alloc() not checking the returned
channel pointer for failure.
* Fixed some potential chan=NULL pointer usage in func_odbc.c. Protected
by testing the bogus_chan value.
* Fixed needlessly clearing a 1024 char auto array when setting the first
char to zero is enough in manager.c:action_getvar().
(closes issue ASTERISK-18613)
Reported by: Thomas Arimont
Patches:
jira_asterisk_18613_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: Thomas Arimont
Russell Bryant [Fri, 23 Sep 2011 00:44:19 +0000 (00:44 +0000)]
Comment out entries in sample res_pktccops.conf.
With these options enabled, they can cause Asterisk to freak out by
SYN flooding a network and eating the CPU. Obviously it would be good to
fix the code so that this can't happen, but we can at least change the default
configuration so it doesn't happen.
This was reported downstream to the Fedora issue tracker:
Richard Mudgett [Thu, 22 Sep 2011 21:29:46 +0000 (21:29 +0000)]
Made ISDN not add numbering plan prefix strings to empty numbers.
When the Caller-ID is restricted, the expected behavior is for the
Caller-ID to be blank. In chan_dahdi, the national prefix is placed onto
the Caller-ID number even if it is restricted (empty) causing the
Caller-ID to be the national prefix rather than blank.
This behavior was lost when sig_pri was extracted from chan_dahdi.
* Made not add prefix strings to empty connected line, calling, and ANI
number strings.
Matthew Jordan [Tue, 20 Sep 2011 22:38:54 +0000 (22:38 +0000)]
Fix for incorrect voicemail duration in external notifications
This patch fixes an issue where the voicemail duration was being reported
with a duration significantly less than the actual sound file duration.
Voicemails that contained mostly silence were reporting the duration of
only the sound in the file, as opposed to the duration of the file with
the silence. This patch fixes this by having two durations reported in
the __ast_play_and_record family of functions - the sound_duration and the
actual duration of the file. The sound_duration, which is optional, now
reports the duration of the sound in the file, while the actual full duration
of the file is reported in the duration parameter. This allows the voicemail
applications to use the sound_duration for minimum duration checking, while
reporting the full duration to external parties if the voicemail is kept.
(issue ASTERISK-2234)
(closes issue ASTERISK-16981)
Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH
Tested by: Matt Jordan
Kinsey Moore [Tue, 20 Sep 2011 21:04:11 +0000 (21:04 +0000)]
Make CANMATCH with the new pattern match engine behave more like the old one
When checking an extension for E_CANMATCH using the new extension matching
algorithm, an exact match was not returned as a possible match resulting in the
queue failing to allow a caller to exit on DTMF. This removes the requirement
that an extension be longer than acquired digits for an E_CANMATCH operation
to succeed.
Richard Mudgett [Tue, 20 Sep 2011 19:10:30 +0000 (19:10 +0000)]
Check if a channel was created before using the pointer in sig_ss7_new_ast_channel().
Fixes the crash in ASTERISK-17955 gdb-11918.txt backtrace.
* Added some missing libss7 access lock protection.
* Prevent cancelling the ss7_linkset() thread at inoportune times just
like the pri_dchannel() thread.
(issue ASTERISK-17955)
Reported by: Ian M Sherman
Patches:
jira_asterisk_17955_v1.8.patch (license #5621) patch uploaded by rmudgett
(attached to related ASTERISK-17966)
Russell Bryant [Tue, 20 Sep 2011 00:56:20 +0000 (00:56 +0000)]
Fix crashes in ast_rtcp_write().
This patch addresses crashes related to RTCP handling. The backtraces just
show a crash in ast_rtcp_write() where it appears that the RTP instance is no
longer valid. There is a race condition with scheduled RTCP transmissions and
the destruction of the RTP instance. This patch utilizes the fact that
ast_rtp_instance is a reference counted object and ensures that it will not get
destroyed while a reference is still around due to scheduled RTCP
transmissions.
RTCP transmissions are scheduled and executed from the chan_sip scheduler
context. This scheduler context is processed in the SIP monitor thread. The
destruction of an RTP instance occurs when the associated sip_pvt gets
destroyed (which happens when the sip_pvt reference count reaches 0). However,
the SIP monitor thread is not the only thread that can cause a sip_pvt to get
destroyed. The sip_hangup function, executed from a channel thread, also
decrements the reference count on a sip_pvt and could cause it to get
destroyed.
While this is being changed anyway, the patch also removes calling
ast_sched_del() from within the RTCP scheduler callback. It's not helpful.
Simply returning 0 prevents the callback from being rescheduled.
(closes issue ASTERISK-18570)
Related issues that look like they are the same problem:
Various changes to allow 1.8 to compile on Mac OS X Lion (10.7)
* Makefile workaround for 10.6 extended to work on 10.7 and later.
* Now uses the 'weak' symbol for Lion systems, which no longer support
'weak_import'
Jonathan Rose [Mon, 19 Sep 2011 20:07:36 +0000 (20:07 +0000)]
Document applications that play audio and do not answer unanswered calls.
This patch is part of an effort to document early media and its usage. If you are
interested in contributing to this documentation effort, there are probably other
applications worth documenting as well as an Asterisk wiki article at
https://wiki.asterisk.org/wiki/display/AST/Early+Media+and+the+Progress+Application
Richard Mudgett [Mon, 19 Sep 2011 18:46:40 +0000 (18:46 +0000)]
Made Dial d and H options no longer immediately auto-answer the calling leg.
The Dial d and H options break DTMF attended transfer atxferdropcall
option.
1) Party A calls party B.
2) Party B does a DTMF attended transfer to Party C.
If the dialplan uses the Dial d or H options to call Party C then the Dial
application answers the call immediately before initiating the call leg to
Party C. The premature answer causes the transfer code to not invoke the
atxferdropcall=no behavior for a blonde transfer since Party C has
"answered". The transfer code thinks that Party B has "consulted" with
Party C when Party B hangs up and completes the transfer to Party A.
Party A now hears ringback until Party C actually answers.
ASTERISK-13294 Dial d option.
ASTERISK-11067 Dial H option to disconnect before answer.
The referenced issues made Dial answer with the d and H options because
many SIP and ISDN phones cannot send DTMF before the call is connected.
* Made require the dialplan to control when or if the call needs to be
answered to use the Dial application d and H options. (The call is no
longer surprise answered when using the Dial d or H options.)
Recently iLBC support in Asterisk has changed after the acquisition of GIPS
by Google. More information about how this may affect you is available in a
blog post at:
Jonathan Rose [Fri, 16 Sep 2011 19:53:40 +0000 (19:53 +0000)]
Fix bad RTP media bridges in directmedia calls on peers separated by multiple Asterisk nodes.
In a situation involving devices on separate Asterisk trunks, the remote RTP bridge would
break when starting a call with directmedia. This patch queues a new type of control frame
so that our RTP bridge loop can properly detect when these situations occur and check to see
if peers need to be updated in order to send their media to the proper location.
a channel lock must never be held with the queues container lock held.
the deadlock occured on masquerade.
the queues container lock is a relic of the past the old queue module lock.
with ao2 there is no need to hold this lock when dealing with members this
patch removes unneeded locks.
(closes issue ASTERISK-18101)
(closes issue ASTERISK-18487)
Reported by: Paul Rolfe, Jason Legault
Tested by: irroot, Jason Legault, Paul Rolfe
Reviewed by: Matthew Nicholson
Richard Mudgett [Wed, 14 Sep 2011 18:21:35 +0000 (18:21 +0000)]
Remove unnecessary libpri dependency checks in the configure script.
Using the --with-pri option with the configure script generated an error
about not having PRI_L2_PERSISTENCE if you did not have the absolute
latest libpri SVN checkout installed.
The AST_EXT_LIB_SETUP_DEPENDENT macro in the configure.ac script seems to
be for libraries that are dependent upon other libraries and not
necessarily for optional/added features within a library.
(closes issue ASTERISK-18535)
Reported by: Michael Keuter
Do parse the option "defaultlanguage" from the [options] section of
asterisk.conf, as in the sample config file. Otherwise the build-time
default language (normally "en") is always the default one.
projects / thirdparty / asterisk.git / log
