Age | Commit message (Collapse) | Author |
|
When a static realtime peer with qualify=yes is loaded, Asterisk will fail to
send an OPTIONS request due to the lastms being equal to 0. This results in
the peer being unable to receive calls from Asterisk because the status is
permanently UNKNOWN.
This patch allows an OPTIONS request to be sent during module load by
ignoring the lastms value on startup only.
Review: https://reviewboard.asterisk.org/r/3294/
(closes issue ASTERISK-17523)
Reported by: Maciej Krajewski
Tested by: wushumasters
patches:
realtime_fix_11.7.0.txt uploaded by Trevor Peirce (license 6112)
........
Merged revisions 410105 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 410106 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 410107 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410108 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
ast_sorcery_objectset_json_create().
* Made exit a loop early on error in ast_sorcery_objectset_json_create().
* Removed some dead code in ast_sorcery_objectset_create2().
........
Merged revisions 410089 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
I observed a crash in res_musiconhold on an Asterisk 11 system using realtime
MOH. Investigation of the backtrace showed a corrupt mohclass, implying that
it got destroyed before the code expected it to. I went looking for reference
counting errors that could have caused this crash and this patch this result.
It contains 2 changes.
1) Remove a usless block of code that was impossible to reach. There was even
a comment indicating that it was impossible to reach. The conditional includes
"!ast_test_flag(global_flags, MOH_CACHERTCLASSES)" and it's inside of an if
block with the opposite check "ast_test_flag(global_flags,
MOH_CACHERTCLASSES)". There's no good reason to keep it around.
2) A similar block to #1 contained a reference counting error. It stores
state->class in the local variable mohclass without increasing its reference
count. The reference count on mohclass is decremented at the end of the
function. This block of code probably very rarely runs, which would help
explain why this system was working fine for many months before experiencing a
crash.
Review: https://reviewboard.asterisk.org/r/3282/
........
Merged revisions 410043 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 410044 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 410090 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410091 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This patch creates the AST_SORCERY dialplan function which allows someone to
retrieve any value from a sorcery-based config file. It's similar to
AST_CONFIG.
The creation of the function itself was fairly straightforward but it required
changes to the underlying sorcery infrastructure that rippled into individual
sorcery objects. The changes stemmed from inconsistencies in how sorcery
created ast_variable objectsets from sorcery objects and the inconsistency
in how individual objects used that feature especially when it came to
parameters that can be specified multiple times like contact in aor and match
in identify. You can read more here...
http://lists.digium.com/pipermail/asterisk-dev/2014-February/065202.html
So, what this patch does, besides actually creating the AST_SORCERY function,
is the following...
* Creates ast_variable_list_append which is a helper to append one ast_variable
list to another.
* Modifies the ast_sorcery_object_field_register functions to accept the
already-defined sorcery_fields_handler callback.
* Modifies ast_sorcery_objectset_create to accept a parameter indicating return
type preference...a single ast_variable with all values concatenated or an
ast_variable list with multiple entries. Also fixed a few bugs.
* Modifies individual sorcery object implementations to use the new function
definition of the ast_sorcery_object_field_register functions.
* Modifies location.c and res_pjsip_endpoint_identifier_ip.c to implement
sorcery_fields_handler handlers so they return multiple occurrences as an
ast_variable_list.
* Added a whole bunch of tests to test_sorcery.
(closes issue ASTERISK-22537)
Review: http://reviewboard.asterisk.org/r/3254/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Transport TOS values were interpreted as DSCP values without being documented
as such. Endpoint TOS values (tos_audio/tos_video) behaved normally as TOS
values have historically. This patch makes the transport TOS values behave as
TOS values and makes all TOS values readable as string values (e.g. AF11).
In addition, alembic scripts have been updated to use the proper field types
for all TOS/COS values.
(issue ASTERISK-23235)
Reported by: George Joseph
Review: https://reviewboard.asterisk.org/r/3304/
........
Merged revisions 410028 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410029 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This change adds a target_uri field to the live recording object. It
contains the URI of what is being recorded.
(closes issue ASTERISK-23258)
Reported by: Ben Merrills
Review: https://reviewboard.asterisk.org/r/3299/
........
Merged revisions 410025 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410027 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
not aggregate MWI.
Attempting to link a NULL object into an ao2 container had been benign previously, but since
enabling DO_CRASH in the testsuite, this is now causing a crash. It's better to be right
here anyway.
........
Merged revisions 410011 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410012 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
sorcery: Create AST_SORCERY dialplan function.
This patch creates the AST_SORCERY dialplan function which allows someone to
retrieve any value from a sorcery-based config file. It's similar to
AST_CONFIG.
The creation of the function itself was fairly straightforward but it required
changes to the underlying sorcery infrastructure that rippled into individual
sorcery objects. The changes stemmed from inconsistencies in how sorcery
created ast_variable objectsets from sorcery objects and the inconsistency
in how individual objects used that feature especially when it came to
parameters that can be specified multiple times like contact in aor and match
in identify. You can read more here...
http://lists.digium.com/pipermail/asterisk-dev/2014-February/065202.html
So, what this patch does, besides actually creating the AST_SORCERY function,
is the following...
* Creates ast_variable_list_append which is a helper to append one ast_variable
list to another.
* Modifies the ast_sorcery_object_field_register functions to accept the
already-defined sorcery_fields_handler callback.
* Modifies ast_sorcery_objectset_create to accept a parameter indicating return
type preference...a single ast_variable with all values concatenated or an
ast_variable list with multiple entries. Also fixed a few bugs.
* Modifies individual sorcery object implementations to use the new function
definition of the ast_sorcery_object_field_register functions.
* Modifies location.c and res_pjsip_endpoint_identifier_ip.c to implement
sorcery_fields_handler handlers so they return multiple occurrences as an
ast_variable_list.
* Added a whole bunch of tests to test_sorcery.
(closes issue ASTERISK-22537)
Review: http://reviewboard.asterisk.org/r/3254/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
When acting as a T.38 fax gateway, res_fax_spandsp would at times cause a crash
in libspandsp. This would occur when, during fax tone detection, a ulaw/alaw
frame would be passed to modem_connect_tones_rx. That particular routine
expects the data to be in slin format. This patch looks at the frame type and,
if the data is ulaw/alaw, converts the format to slin before passing it to
modem_connect_tones_rx.
Review: https://reviewboard.asterisk.org/r/3296
(closes issue ASTERISK-20149)
Reported by: Alexandr Gordeev
Tested by: Michal Rybarik
patches:
spandsp_g711decode.diff uploaded by Michal Rybarik (license 6578)
........
Merged revisions 409990 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 409991 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409996 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
Merged revisions 409976 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409977 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
* Made the moh_register() define use useful parameter names.
........
Merged revisions 409967 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409970 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
The test of the result of the stat() call was inverted such that its
output was only used if the call failed. This inverts the test so that
the output of stat() is used correctly. This was causing full reloads
on unchanged files.
(closes issue ASTERISK-23383)
Reported by: David Woolley
........
Merged revisions 409916 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 409917 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 409918 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409919 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
It is possible for a channel to be masqueraded out of a bridge which
means it may no longer have RTP glue to check upon leaving said bridge.
If this situation occurred (it's possible at least during dial and call
pickup) then Asterisk would crash. This change makes sure the glue is
checked before use.
(closes issue AST-1290)
Reported by: John Bigelow
........
Merged revisions 409900 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409904 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
* Added the queues and queue_members tables to the config alembic scripts.
* Added the CDR table alembic creation script. The CDR table is more of
an example for new setups since the actual table can be fully customized
in cdr_adaptive_odbc.conf.
(closes issue ASTERISK-23233)
Reported by: jmls
Review: https://reviewboard.asterisk.org/r/3227/
........
Merged revisions 409885 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409889 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
presence hint.
There was a missing comma.
This was discovered by Dan Kaplan.
........
Merged revisions 409886 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 409887 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409888 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
When nanosecond time resolution was added for identifying config file
changes, it didn't cover all of the myriad of ways that one might obtain
nanosecond time resolution off of struct stat.
Rather than complicate the #if even further figuring out one system from
the next, this patch directly tests for the three struct members I know
about today, and #ifdef's accordingly.
Review: https://reviewboard.asterisk.org/r/3273/
........
Merged revisions 409833 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 409834 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 409835 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409836 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
format for uint64_t
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409832 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Several fixes for the WebSockets implementation in res/res_http_websocket.c
* Flush the websocket session FILE* as fwrite() may not actually guarantee sending
the data to the network. If we do not flush, it seems that buffering on the SSL
socket for outbound messages causes issues
* Refactored ast_websocket_read to take into account that SSL file descriptors
may be ready to read via fread() but poll() will not actually say so because
the data was already read from the network buffers and is now in the libc buffers
(closes issue ASTERISK-23099)
(closes issue ASTERISK-21930)
Review: https://reviewboard.asterisk.org/r/3248/
........
Merged revisions 409681 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 409697 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409831 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
Merged revisions 409777 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 409778 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 409779 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409780 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
Correct RTP handling in chan_unistim and fix transfer process broken in previous fix:
- Fixed too early RTP setup with phone, that cause no ringback tone on caller side
- Handle call transfer cancel only in STATE_CALL case (related to ASTERISK-23073)
(Reported by: Németh Tamás, niurkin sil)
........
Merged revisions 409761 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409763 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
functions conforming to other channel drivers. Do not forget auto-detected and user-selected phone settings on 'unistim reload'
........
Merged revisions 409705 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 409745 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409747 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
Merged revisions 409682 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409683 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This patch prevents a crash when using the function audiohookinheritance without
setting the channel.
(closes issue ASTERISK-23104)
Reported by: Joel Vandal
Tested by: Joel Vandal
Patches:
asterisk-23104_audiohook_inherit_no_channel-11.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/3272/
........
Merged revisions 409623 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 409625 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 409626 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
ICE sessions will now be restarted if sessions are changed to use new sets of
remote candidates.
(closes issue ASTERISK-22911)
Reported by: Vytis Valentinavičius
Review: https://reviewboard.asterisk.org/r/3275/
........
Merged revisions 409565 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 409570 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409587 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This adds an assert that will only be active if Asterisk is compiled
with DO_CRASH and allows the testsuite to fail tests that would
otherwise require log file parsing.
........
Merged revisions 409566 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 409567 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 409568 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409569 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
(closes issue ASTERISK-23406)
Reported by: ibercom
Tested by: ibercom
Patches:
asterisk-11.patch uploaded by ibercom
........
Merged revisions 409472 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 409473 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 409474 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409475 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
Merged revisions 409422 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
Merged revisions 409361 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 409362 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 409363 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409364 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
-O6 is not a legal option of gcc. Unofficially gcc considers it to be
equivalent of -O3. clang chalks on it, though. This commit sets the
default optimization flag to be -O3, like gcc actually considered it.
Review: https://reviewboard.asterisk.org/r/3280/
........
Merged revisions 409308 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 409344 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 409346 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409350 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This change passes options to the UAS creation function. This in turn
sets up 100rel and session timer properties on the incoming session.
Reported by Julian Russell on asterisk-users mailing list.
........
Merged revisions 409287 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409288 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
Merged revisions 409274 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409275 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
Merged revisions 409272 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409273 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
* Remove some unnecessary RAII_VAR() usage.
* Made the struct stasis_subscription ao2 object use the ao2 lock instead
of a redundant join_lock in the struct for ast_cond_wait().
* Removed locks on some ao2 objects that don't need the lock.
* Made the topic pool entries container use the ao2 template functions.
* Add some missing allocation failure checks.
* Add missing cleanup in off nominal path of dispatch_message().
........
Merged revisions 409270 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409271 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
* Add precautionary p->owner checks in sip_hangup(), get_refer_info(),
get_also_info(), and interpret_t38_parameters().
* Simplify some tangled logic in get_refer_info(), get_also_info(), and
add_rpid().
* Removed some dead code in handle_request_invite().
(closes issue ASTERISK-23323)
Reported by: Walter Doekes
Patches:
issueA23323-more_p_owner_checks-1.8.x.patch (license #5674) uploaded by wdoekes (modified)
issueA23323-more_p_owner_checks-11.x.patch (license #5674) uploaded by wdoekes (modified)
issueA23323-more_p_owner_checks-12.x.patch (license #5674) uploaded by wdoekes (modified)
issueA23323-more_p_owner_checks-trunk.patch (license #5674) uploaded by wdoekes (modified)
........
Merged revisions 409207 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 409255 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 409256 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409257 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
During the rewrite of AMI events to use the Stasis bus, the name of the
QueueMemberPaused event was changed to QueueMemberPause. This corrects
documentation to reflect that.
........
Merged revisions 409234 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409237 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
* Fix crash in ast_channel_hangupcause_set() because p->owner not checked
before calling. Regression introduced by the fix for ASTERISK-22621.
(closes issue ASTERISK-23135)
Reported by: OK
(issue ASTERISK-23323)
Reported by: Walter Doekes
........
Merged revisions 409156 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 409157 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 409158 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409159 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
r409129 | jrose | 2014-02-27 13:19:02 -0600 (Thu, 27 Feb 2014) | 15 lines
res_rtp_asterisk: Fix checklist creating problems in ICE sessions
Prior to this patch, local candidate lists including SRFLX would fail to start
properly when building ICE candidate check lists. This patch fixes that problem
by making sure that each SRFLX candidate is associated with the proper
base address so that the check list can create matches properly.
This patch was written by jcolp. The issue will be left open to await testing
by the issue participants.
(issue ASTERISK-23213)
Reported by: Andrea Suisani
Review: https://reviewboard.asterisk.org/r/3256/
........
r409130 | jrose | 2014-02-27 13:38:10 -0600 (Thu, 27 Feb 2014) | 8 lines
res_rtp_asterisk: correct build error from r409129
Accidentally placed a declaration below functional code
(issue ASTERISK-23213)
Reported by: Andrea Suisani
Review: https://reviewboard.asterisk.org/r/3256/
........
Merged revisions 409129-409130 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 409131 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409132 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This memset complained in dev mod on my Ubuntu box. The memset is both
unnecessary and dangerous. At this point, m hasn't been initialized
yet, so the memset will write off to whatever address happens to be
on the stack at the time.
........
Merged revisions 409077 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 409083 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 409087 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409091 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Comment out many settings in res_fax.conf.sample. The defaults are set in
res_fax.c, so setting the same value in sample config does nothing but make
the sample config more fragile.
(closes issue ASTERISK-23231)
Reported by: David Brillert
Review: https://reviewboard.asterisk.org/r/3261/
........
Merged revisions 409052 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 409053 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 409054 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409055 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
The setting 'use_ptime' is supposed to tell Asterisk to honour the ptime
attribute in an offer, preferring it to whatever packetization
preferences have been set internally. Currently, however, something
rather quirky will happen:
(1) The SDP answer will be constructed in create_outgoing_sdp_stream.
This will use the preferences from the endpoint, such that the 200 OK
response will add the packetization preferences from the endpoint, and
not what was offered.
(2) When the 200 response is issued, apply_negotiated_sdp_stream is called.
This will call apply_packetization, which will use the ptime attribute
from the offer internally.
We end up telling the offerer to use the internal ptime attribute, but we end
up using the offered ptime attribute. Hilarity ensues.
This patch modifies the behaviour by calling apply_packetization from
negotiate_incoming_sdp_stream, which is called prior to
create_outgoing_sdp_stream. This causes the format preferences on the
session's media object to be set to the inbound ptime value (if 'use_ptime'
is enabled), such that the construction of the answer gets the right value
immediately.
Review: https://reviewboard.asterisk.org/r/3244/
........
Merged revisions 408999 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409000 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
* Make the consumer ao2 object use the ao2 lock instead of a redundant
lock in the struct for ast_cond_wait().
* Fixed some curly brace placements.
* Fixed use of malloc(0). malloc(0) has variant behavior. It is up to
the implementation to determine if it returns NULL or a valid pointer that
can be later passed to free().
........
Merged revisions 408983 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408984 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
When accidentally compiling against a wrong version of
pjsip headers with a different pjsip_inv_session size,
the invite_tsx structure could be null in the answer()
function. This led to a crash because it attempted to
send the session response with an uninitialized packet
pointer. This patch presets packet to null and adds a
diagnostic log message to explain why the call fails.
Review: https://reviewboard.asterisk.org/r/3267/
........
Merged revisions 408970 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408971 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
the system.
This change makes some error cases use ast_ari_response_error to construct their
error responses instead of manually doing it. This ensures they are consistent
with the other error responses.
Based on the original patch as done by Paul Belanger on the associated review.
Review: https://reviewboard.asterisk.org/r/2904/
........
Merged revisions 408957 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408958 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
Merged revisions 408943 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408944 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
It is currently possible for an ast_sip_session to exist without an
associated channel as is the case when a new invite is coming in or
just after a hangup is issued on a chan_pjsip channel. Part of the
attended transfer code assumed the channel would be non-NULL and used
it as such causing a crash. This bug was exposed thanks to the attended
transfer ARI test in the test suite.
(closes issue ASTERISK-23287)
Reported by: Matt Jordan
........
Merged revisions 408941 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408942 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
support.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408932 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Added presence support for digium phones.
Review: https://reviewboard.asterisk.org/r/3239/
........
Merged revisions 408882 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408883 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Added the ability for transferring directly to voicemail on digium phones.
Added a new module that checks for the presence of a custom header and/or
diversion header within a sip REFER. If either is found and they specify
a sending to voicemail action then variables are added to the channel
allowing the user access to them in the dialplan. Dialplan can then be
written that branches based upon these values allowing, for instace, for
a single number to be used for dialing and/or accessing voicemail directly.
Also fixed a problem where the PJSIP_HEADER function was allowing non pjsip
channels through (checked to make sure it has the correct channel type before
proceeding).
Review: https://reviewboard.asterisk.org/r/3245/
........
Merged revisions 408880 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408881 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Made the wording a bit more explicit. Didn't really change the meaning.
........
Merged revisions 408876 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 408877 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 408878 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408879 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
It is possible to pre-load pbx_config. As a result, pbx_config - which will
load and parse the dialplan - will attempt to use various dialplan components,
such as device state providers and presence state providers, prior to them
being initialized by the core. This would lead to a crash, as the components
had not created their Stasis cache entries.
This patch moves a number of core component initializations before the module
pre-load. This guarantees that if someone does pre-load pbx_config - or other
pbx modules - that the Stasis caches for the various core components are
created.
(closes issue ASTERISK-23320)
Reported by: xrobau
(closes issue ASTERISK-23265)
Reported by: Andrew Nagy
Tested by: Andrew Nagy, Rusty Newton
........
Merged revisions 408855 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408859 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|