Age | Commit message (Collapse) | Author |
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Make the comments follow doxygen format, move comments to the line
before each field they describe.
Change-Id: Ic445468398b5e88f13910f7c2f70bd15aad33a27
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* changes:
acl: Fix allocation related issues.
acl: Update logger message to match 15+.
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out." into 13
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Mac doesn't like the comparison of -1 to an enum, so store the result of
ast_sip_str_to_dtmf to an int so we can check for the negative return
value. ast_sip_str_to_dtmf returns an int so this is only delaying the
implicit type cast.
Change-Id: I0c262c1719ee951aae1f437d733a301cf5f8ad29
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test_pbx used raise without explicitly including signal.h. On Mac for
some reason nothing else includes it.
test_logger checked if an unsigned int was negative. Switch the
variable to 'int' so that error check can be effective.
Change-Id: Ie1db5dd1818ac25cc2ae41b644f848b5865b1362
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Some net-snmp builds do not provide the RONLY declare, only
NETSNMP_OLDAPI_RONLY. Map RONLY to NETSNMP_OLDAPI_RONLY to get around
this error.
Change-Id: Ida5c7ad9406515825485c4d3b4a34fd6ad0da577
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It's impossible for gwtimeout or fdtimeout to be less than 0 because
they are unsigned int's. Remove checks and unreachable branches.
Change-Id: Ib2286960621e6ee245e40013c84986143302bc78
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Declare 'res' initialized to -1 to deal with earlier error paths that
could cause 'res' to be returned uninitialized.
Change-Id: I8ac2a5755bf4174d89ef893e924c940f702b104e
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Remove doxygen contents that have nothing to do with the current state
of Asterisk.
Change-Id: Ic072cc8641f9533a202990ccf275ce87e3efd95c
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Domains themselves can be up to 255 characters long (per RFC 1035), so
our current buffer sizes are wholly inadequate for many use cases.
Change-Id: If3f30a68307f1365a1fe06bc4b854c62842c9292
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into 13
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Add checks for allocation errors, cleanup and report failure when they
occur.
* ast_duplicate_acl_list: Replace log warnings with errors, add missing
line-feed.
* ast_append_acl: Add missing line-feed to logger message.
* ast_append_ha: Avoid ast_strdupa in loop by moving debug message to
separate function.
* ast_ha_join: Use two separate calls to ast_str_append to avoid using
ast_strdupa in a loop.
Change-Id: Ia19eaaeb0b139ff7ce7b971c7550e85c8b78ab76
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This patch causes a logger message to be the same as it is in 15+. This
will allow a follow-up patch to be cherry-picked to all 3 branches.
Change-Id: Ic0665a3d49987e4eb6df28dcd9e90b1c3ca191e0
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This is a fun one.
Given the following attended transfer scenario:
1. Transfer target is called
2. Transferer hangs up
3. Transfer target call attempt reaches timeout
4. Transfer target is told to hang up
5. Transfer target answers before channel is hung up
6. Transferer recall target is called
A crash would occur. This is because the transfer target call
attempt, despite being told to hang up, would raise a recall
target answer before the recall target had been answered. As it
had not answered there would be no recall target channel and it
would implode.
This change makes it so that if the transfer target has been
hung up we don't tell the attended transfer code that it has
answered. We also clear out the stimulus that the recall target
has been answered after telling the transfer target to hang up,
in case it was able to raise the information before we told it
to hangup.
ASTERISK-27361
Change-Id: Ifb8b255a9c4d2c5c1b8ad77bf54f659ed286df99
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aoc_publish_blob failed to check for msg allocation error and never
released msg.
Change-Id: Ib31a9ffb81056a0d496a49d7eec795005a44bcd5
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We were not \0 terminating this string, so any attempt to print it would
in the best case show an empty string and in the worst case potentially
crash.
Change-Id: I63d96ef8f7516ac02a0f91e22dfa8acdc615042c
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This improves performance for registrations assuming that
res_config_astdb is not in use.
Change-Id: I86f37aa9ef07a4fe63448cb881bbadd996834bb1
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* Balanced the session->inv_session refs on answer failure.
Change-Id: I33542d639d37e692cb46550b972a5fcfc3b804b8
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Memory corruption happened to the media frame caches when an audio hook
freed a frame when it shouldn't. I think the freed frame was because a
jitter buffer interpolated a missing frame and the audio hook
unconditionally freed it.
* Made audiohook.c:audio_audiohook_write_list() not free an interpolated
frame if it is the same frame as what was passed into the routine.
* Made plc.c:normalise_history() use memmove() instead of memcpy() on a
memory block that could overlap. Found by valgrind investigating this
issue.
ASTERISK-27238
ASTERISK-27412
Change-Id: I548d86894281fc4529aefeb9f161f2131ecc6fde
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We've been calling pbx_builtin_setvar_helper to set the
RECORD_STATUS variable before actually closing the recorded file.
If a client is watching VarSet events and tries to do something with
the file when a RECORD_STATUS event is seen, they might attempt to
do so while the file it's still open.
We now delay calling pbx_builtin_setvar_helper until after we close
the file.
ASTERISK-27423
Change-Id: I7fe9de99953e46b4bafa2b38cf151fe8f6488254
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The OUTPUTDIR environment variable can now be set either in the
environment itself or in ast_debug_tools.conf. If set, it's used
for all work products instead of /tmp.
Also added the --tarball-config option that includes the contents
of /etc/asterisk when either --tarball-coredumps or --tarball-results
are used.
Change-Id: I66b2553319df61caea5b313d084f51978f730b4c
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into 13
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Previously for PJSIP the local address of WebSocket connections
was set to the remote address. For logging purposes this is
not particularly useful.
The WebSocket API has been extended to allow the local
address to be queried and this is used in PJSIP to set the
local address to the correct value.
The PJSIP HEP support has also been tweaked so that reliable
transports always use the local address on the transport
and do not try to (wrongly) guess. As they are connection
based it is impossible for the source to be anything else.
ASTERISK-26758
ASTERISK-27363
Change-Id: Icd305fd038ad755e2682ab2786e381f6bf29e8ca
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Change-Id: I4ea49c441890a81384144479dc93ab5a3989486d
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The remote console socket path is the combination of asterisk.conf
settings astrundir from [directories] and astctl from [files].
Unconditionally combine the two strings after processing all values
to ensure we end up with the correct socket path.
ASTERISK-27415
Change-Id: Ib1e2805d55d6b0955c6430a1a2a93acbf9b091e8
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The default return code for pjsip_find_msg was PJ_SUCCESS so if
a Content-Length header wasn't found at all, pjsip_find_msg was
returning PJ_SUCCESS instead of PJSIP_EMISSINGHDR.
Also added the volatile keyword to a few variables that are used
both inside and outside the PJ_TRY/PJ_CATCH block.
Partial fix for ASTERISK_27408
Change-Id: If82ba9de921e3d57df9c68cf96ee45ccc1491f7a
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Update from 2.7 to 2.7.1 for bundled pjproject. Changed version
and removed patch files included in the update.
Change-Id: I55cea8e734b318c2df9daf86aa0802c559ec8357
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Some consumers of the sorcery API use ast_sorcery_retrieve_by_regex
only so that they can anchor the potential match as a prefix and not
because they truly need regular expressions.
Rather than using regular expressions for simple prefix lookups, add
a new operation - ast_sorcery_retrieve_by_prefix - that does them.
Patches against 13 and 15 have a compatibility layer needed to
maintain ABI that is not needed in master.
Change-Id: I56f4e20ba1154bd52281f995c27a429a854f6a79
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* AST_VECTOR_STEAL_ELEMENTS - steal the array of elements for use
with non-vector code.
* struct ast_vector_string - a vector of 'char *'.
Change-Id: I104d1b204be03fccf67e02a195596adcb5ab1e42
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This change causes the configure script to fail if the C compiler does
not support both function attributes constructor and destructor. These
were already required as modules cannot function without these attributes
and Asterisk requires modules.
This also has AST_GCC_ATTRIBUTE set a variable
ax_cv_have_func_attribute_$1. This is the same variable name used by
autoconf-archive's AX_GCC_FUNC_ATTRIBUTE, used for the same purpose.
Change-Id: Id68e8a1447f2a6d707c54b56350e7bfdb33fb663
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