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When ast_config_load() fails with CONFIG_STATUS_FILEINVALID, it has
already destroyed the ast_config struct for us. Trying to do it again
results in a crash.
Change-Id: If6a5c0ca718ad428e01a1fb25beb209a9ac18bc6
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ASTERISK-26802 #close
Reported by: Michael L. Young
Change-Id: Iad293080f55d4d69ab615717a15211d916eed613
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* Fix tcptls_session ref and fd leak in ast_tcptls_server_root().
Change-Id: I0ddf01cd3c10d3b6666d7bf68d4e206a37f4fbdb
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When AMI encounters an error at the beginning of a session, it would
explicitly call ast_iostream_close() on its tcptls session's iostream.
It then would jump to a label where it would shut down the tcptls
session instance. The tcptls session instance would again attempt to
close the iostream.
Under normal circumstances, this might go by unnoticed. However, when
MALLOC_DEBUG is enabled, all fields on the iostream get set to
0xdeaddead when the iostream is freed. Thus a second call to
ast_iostream_close() after the iostream has been freed would reslt in an
attempt to call SSL_shutdown on 0xdeaddead, which would crash and burn
horribly.
The fix here is to not directly close the iostream from the dangerous
scenarios. The specific scenarios are:
* Exceeding the configured authlimit
* Failing to build a mansession on a new connection
Change-Id: I908f98d516afd5a263bd36b072221008a4731acd
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This creates the following:
* Asterisk's internal representation of an SDP
* An API for translating SDPs from one format to another
* An implementation of a translator for PJMEDIA
Change-Id: Ie2ecd3cbebe76756577be9b133e84d2ee356d46b
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This is step one of adding an SDP API: defining some
configurable settings for SDPs. This is based on options
that are currently supported in Asterisk.
Change-Id: I1ede91aafed403b12a9ccdfb91a88389baa7e5d7
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On some platforms a multiarch approach is used for libraries.
The build system does not take this into account and still
places libraries into the lib directory if no --libdir is
specified to configure. On initial startup this results in
libasteriskssl.so not being found, as it is not in the multiarch
lib directory.
This change does the minimally invasive thing and executes
ldconfig so that the libraries in the lib directory are found
and their location cached. By doing so Asterisk starts up fine.
ASTERISK-26705
Change-Id: I6d30b6427e9d5e69470e11327c7ff203fa7da519
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The realtime framework allows for components to look up values using a
LIKE clause with similar syntax to SQL's. pbx_realtime uses this
functionality to search for pattern matching extensions that start with
an underscore (_).
When passing an underscore to SQL's LIKE clause, it will be interpreted
as a wildcard matching a single character and therefore needs to be
escaped. It is (for better or for worse) the responsibility of the
component that is querying realtime to escape it with a backslash before
passing it in. Some RDBMs support escape characters by default, but the
SQL92 standard explicitly says that there are no escape characters
unless they are specified with an ESCAPE clause, e.g.
SELECT * FROM table WHERE column LIKE '\_%' ESCAPE '\'
This patch instructs 3 backends - res_config_mysql, res_config_pgsql,
and res_config_sqlite3 - to use the ESCAPE clause where appropriate.
Looking through documentation and source tarballs, I was able to
determine that the ESCAPE clause is supported in:
MySQL 5.0.15 (released 2005-10-22 - earliest version available from
archives)
PostgreSQL 7.1 (released 2001-04-13)
SQLite 3.1.0 (released 2005-01-21)
The versions of the relevant libraries that we depend on to access MySQL
and PostgreSQL will not work on versions that old, and I've added an
explicit check in res_config_sqlite3 to only use the ESCAPE clause when
we have a sufficiently new version of SQLite3.
res_config_odbc already handles the escape characters appropriately, so
no changes were required there.
ASTERISK-15858 #close
Reported by: Humberto Figuera
ASTERISK-26057 #close
Reported by: Stepan
Change-Id: I93117fbb874189ae819f4a31222df7c82cd20efa
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To be consistent with sdp implementation.
Change-Id: I714e300939b4188f58ca66ce9d1e84b287009500
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There were two specific issues resolved here:
1) The code that iterated over the required fields
(via ast_realtime_require) was broken for the RQ_INTEGER1 field
type. Iteration would stop when the first RQ_INTEGER1 (0) field
was encountered.
2) sqlite3_changes() was used to try and count the number of rows
returned by a SELECT statement. sqlite3_changes() only counts
affected rows, so this was always returning the value from the
most recent data modification statement. We now separate read-only
queries from data modification queries and count rows appropriately
in both cases.
ASTERISK-23457 #close
Reported by: Scott Griepentrog
Change-Id: I91ed20494efc3fcfbc2a96ac7646999a49814884
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ASTERISK-26794 #close
Change-Id: I9cbc3b6b6a8aab590f5ccde9c262a98e4d5253a1
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There is difference exists in behaviour of char type on x86 and ARM.
On x86 by default char variable type means signed char, but in ARM
unsigned char used. This make binary calculations and negative values
works wrong on ARM.
This patch change type of char variables used for store negative
values and binary calculations to signed char.
ASTERISK-26714
Change-Id: Id78716dee9568a58419d4ef63c038affc3dfc7ab
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This patch fixes 2 original issues and more that those 2 exposed.
* When we send a NOTIFY, and the client either doesn't respond or
responds with a non OK, pjproject only calls our
pubsub_on_evsub_state callback, no others. Since
pubsub_on_evsub_state (which does the sub_tree cleanup) does not
expect to be called back without the other callbacks being called
first, it just returns leaving the sub_tree orphaned. Now
pubsub_on_evsub_state checks the event for PJSIP_EVENT_TSX_STATE
which is what pjproject will set to tell us that it was the
transaction that timed out or failed and not the subscription
itself timing our or being terminated by the client. If is
TSX_STATE, pubsub_on_evsub_state now does the proper cleanup
regardless of the state of the subscription.
* When a client renews a subscription, we don't update the
persisted subscription with the new expires timestamp. This causes
subscription_persistence_recreate to prune the subscription if/when
asterisk restarts. Now, pubsub_on_rx_refresh calls
subscription_persistence_update to apply the new expires timestamp.
This exposed other issues however...
* When creating a dialog from rdata (which sub_persistence_recreate
does from the packet buffer) there must NOT be a tag on the To
header (which there will be when a client refreshes a
subscription). If there is one, pjsip_dlg_create_uas will fail.
To address this, subscription_persistence_update now accepts a flag
that indicates that the original packet buffer must not be updated.
New subscribes don't set the flag and renews do. This makes sure
that when the rdata is recreated on asterisk startup, it's done
from the original subscribe packet which won't have the tag on To.
* When creating a dialog from rdata, we were setting the dialog's
remote (SUBSCRIBE) cseq to be the same as the local (NOTIFY) cseq.
When the client tried to resubscribe after a restart with the
correct cseq, we'd reject the request with an Invalid CSeq error.
* The acts of creating a dialog and evsub by themselves when
recreating a subscription does NOT restart pjproject's subscription
timer. The result was that even if we did correctly recreate the
subscription, we never removed it if the client happened to go away
or send a non-OK response to a NOTIFY. However, there is no
pjproject function exposed to just set the timer on an evsub that
wasn't created by an incoming subscribe request. To address this,
we create our own timer using ast_sip_schedule_task. This timer is
used only for re-establishing subscriptions after a restart.
An earlier approach was to add support for setting pjproject's
timer (via a pjproject patch) and while that patch is still included
here, we don't use that call at the moment.
While addressing these issues, additional debugging was added and
some existing messages made more useful. A few formatting changes
were also made to 'pjsip show scheduled tasks' to make displaying
the subscription timers a little more friendly.
ASTERISK-26696
ASTERISK-26756
Change-Id: I8c605fc1e3923f466a74db087d5ab6f90abce68e
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pjsip limits the total number of ICE candidates to PJ_ICE_MAX_CAND,
which is a compile-time constant. Instead of hard-coding 16 when we
enumerate local interfaces, use PJ_ICE_MAX_CAND so that we can
potentially collect more interfaces if the compile time options are
changed.
Tangentially related to ASTERISK~24464
Change-Id: I1b85509e39e33b1fed63c86261fc229ba14bbabd
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Adds the import tool for converting a HRIR database to hrirs.h
ASTERISK-26292
Change-Id: I51eb31b54c23ffd9b544bdc6a09d20c112c8a547
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This change adds unit tests cover the following:
1. That retrieving the first media stream of a specific media
type from a stream topology retrieves the expected media
stream.
2. That setting the native formats of a channel which does
not support streams results in the creation of streams on
its behalf according to the formats of the channel.
3. That setting a stream topology on a channel which supports
streams sets the topology to the provided one.
ASTERISK-26790
Change-Id: Ic53176dd3e4532e8c3e97d9e22f8a4b66a2bb755
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Original patch by John Covert, slight modifications by me.
ASTERISK-17428 #close
Reported by: John Covert
Patches:
app_voicemail.c.patch (license #5512) patch uploaded by
John Covert
Change-Id: Ic3361b0782e5a5397a19ab18eb8550923a9bd6a6
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Adds topology set and get to channel.
ASTERISK-26790
Change-Id: Ic379ea82a9486fc79dbd8c4d95c29fa3b46424f4
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When attempting to use VoiceMailPlayMsg with a realtime data backend
the message is located, but never retrieved. This patch adds the
required RETRIEVE and DISPOSE calls that will fetch the message from
the database (and IMAP storage as well for that matter).
Also, removed extraneous make_file call.
ASTERISK-26723 #close
Change-Id: I1e122dd53c0f3d7faa10f3c2b7e7e76a47d51b8c
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When using Record() with the silence detection feature, the stream is
written out to the given file. However, if only 'silence' is detected,
this file is then truncated to the first second of the recording.
This patch adds the 'u' option to Record() to override that behavior.
ASTERISK-18286 #close
Reported by: var
Patches:
app_record-1.8.7.1.diff (license #6184) patch uploaded by var
Change-Id: Ia1cd163483235efe2db05e52f39054288553b957
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ASTERISK-26775 #close
Change-Id: I86de4b1a699d6edc77fea9b70d839440e4088284
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This change adds unit tests for the various API calls relating
to stream topologies. This includes creation, destruction,
inspection, and manipulation.
Through this a few bugs were uncovered in the implementation:
1. Creating a topology using a format capabilities would fail as
the code considered a return value of 0 from the append stream
function to indicate an error which is incorrect.
2. Not all functions which placed a stream into a topology
set the position on the stream itself.
3. Appending a stream would cause a frack if the position
provided was the last one. This occurred because the existing
stream was queried but the index was outside of what the
vector was currently at for size.
ASTERISK-26786
Change-Id: Id5590e87c8a605deea1a89e53169a9c011d66fa0
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* app_minivm: Use built-in completion facilities to complete optional
arguments.
* app_voicemail: Use built-in completion facilities to complete
optional arguments.
* app_confbridge: Add missing colons after 'Usage' text.
* chan_alsa: Use built-in completion facilities to complete optional
arguments.
* chan_sip: Use built-in completion facilities to complete optional
arguments. Add completions for 'load' for 'sip show user', 'sip show
peer', and 'sip qualify peer.'
* chan_skinny: Correct and extend completions for 'skinny reset' and
'skinny show line.'
* func_odbc: Correct completions for 'odbc read' and 'odbc write'
* main/astmm: Use built-in completion facilities to complete arguments
for 'memory' commands.
* main/bridge: Correct completions for 'bridge kick.'
* main/ccss: Use built-in completion facilities to complete arguments
for 'cc cancel' command.
* main/cli: Add 'all' completion for 'channel request hangup.' Correct
completions for 'core set debug channel.' Correct completions for 'core
show calls.'
* main/pbx_app: Remove redundant completions for 'core show
applications.'
* main/pbx_hangup_handler: Remove unused completions for 'core show
hanguphandlers all.'
* res_sorcery_memory_cache: Add completion for 'reload' argument of
'sorcery memory cache stale' and properly implement.
Change-Id: Iee58c7392f6fec34ad9d596109117af87697bbca
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This change adds the media stream topology definition and API for
accessing and using it.
Some refactoring of the stream was also done.
ASTERISK-26786
Change-Id: Ic930232d24d5ad66dcabc14e9b359e0ff8e7f568
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