Commit Graph

145 Commits

Author SHA1 Message Date
George Joseph 651290a809 BuildSystem: Fix a few issues hightlighted by gcc 6.x
gcc 6.1.1 caught a few more issues.
Made sure the unit tests still pass for the func_env and stdtime
issues.

ASTERISK-26157 #close

Change-Id: I6664d8f34a45bc1481d2a854481c7878b0c1cf8e
2016-06-28 12:40:49 -05:00
George Joseph c7309a5254 chan_unistim: Fix memcpy in get_to_address
A code block only enabled when HAVE_PKTINFO is not defined (FreeBSD)
was using a pointer to a pointer as the destination of a memcpy and a
'&' instead of '*' in the sizeof.

ASTERISK-26138 #close

Change-Id: Id4927ff256c0e470bdf7bcfc025146a2f656e708
2016-06-22 13:31:58 -05:00
Alexander Traud cf79b62778 ast_format_cap_get_names: To display all formats, the buffer was increased.
ASTERISK-25533 #close

Change-Id: Ie1a9d1a6511b3f1a56b93d04475fbf8a4e40010a
2015-11-09 16:58:52 +01:00
Richard Mudgett 1a549ed134 rtp_engine.c: Initial split of payload types into rx and tx mappings.
There are numerous problems with the current implementation of the RTP
payload type mapping in Asterisk.  It uses only one mapping structure to
associate payload types to codecs.  The single mapping is overkill if all
of the payload type values are well known values.  Dynamic payload type
mappings do not work as well with the single mapping because RFC3264
allows each side of the link to negotiate different dynamic mappings for
what they want to receive.  Not only could you have the same codec mapped
for sending and receiving on different payload types you could wind up
with the same payload type mapped to different codecs for each direction.

1) An independent payload type mapping is needed for sending and
receiving.

2) The receive mapping needs to keep track of previous mappings because of
the slack to when negotiation happens and current packets in flight using
the old mapping arrive.

3) The transmit mapping only needs to keep track of the current negotiated
values since we are sending the packets and know when the switchover takes
place.

* Needed to create ast_rtp_codecs_payload_code_tx() and make some callers
use the new function because ast_rtp_codecs_payload_code() was used for
mappings in both directions.

* Needed to create ast_rtp_codecs_payloads_xover() for cases where we need
to pass preferred codec mappings to the peer channel for early media
bridging or when we need to prefer the offered mapping that RFC3264 says
we SHOULD use.

* ast_rtp_codecs_payloads_xover() and ast_rtp_codecs_payload_code_tx() are
the only new public functions created.  All the others were only used for
the tx or rx mapping direction so the function doxygen now reflects which
direction the function operates.

* chan_mgcp.c: Removed call to ast_rtp_codecs_payloads_clear() as doing
that makes no sense when processing an incoming SDP.  We would be wiping
out any mappings that we set for the possible outgoing SDP we sent
earlier.

ASTERISK-25166
Reported by: Kevin Harwell

ASTERISK-17410
Reported by: Boris Fox

Change-Id: Iaf6c227bca68cb7c414cf2fd4108a8ac98bd45ac
2015-08-19 17:09:58 -05:00
Rodrigo Ramírez Norambuena eec010829a AST_MODULE_INFO: Format corrections to the usages of AST_MODULE_INFO macro.
Change-Id: Icf88f9f861c6b2a16e5f626ff25795218a6f2723
2015-05-13 16:34:23 -05:00
Joshua Colp bebf0b9b27 chan_unistim: Fix build failure due to ACL changes.
Change-Id: I57081045c72b9fcf12d5c84493278f9272c31b32
2015-05-05 17:01:31 -03:00
Mark Michelson 11ffcf662f Restrict functionality when ACLs are misconfigured.
This patch has two main purposes:

1) Improve warning messages when ACLs are configured improperly.
2) Prevent misconfigured ACLs from allowing potentially unwanted
traffic.

To acomplish point (2) in most cases, whatever configuration object that
the ACL belonged to was not allowed to load.

The one exception is res_pjsip_acl. In that case, ACLs are their own
configuration object. Furthermore, the module loading code has no
indication that a ACL configuration had a failure. So the tactic taken
here is to create an ACL that just blocks everything.

ASTERISK-24969
Reported by Corey Farrell

Change-Id: I2ebcb6959cefad03cea4d81401be946203fcacae
2015-04-30 10:43:51 -05:00
Matt Jordan 4a58261694 git migration: Refactor the ASTERISK_FILE_VERSION macro
Git does not support the ability to replace a token with a version
string during check-in. While it does have support for replacing a
token on clone, this is somewhat sub-optimal: the token is replaced
with the object hash, which is not particularly easy for human
consumption. What's more, in practice, the source file version was often
not terribly useful. Generally, when triaging bugs, the overall version
of Asterisk is far more useful than an individual SVN version of a file. As a
result, this patch removes Asterisk's support for showing source file
versions.

Specifically, it does the following:

* Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and
  remove passing the version in with the macro. Other facilities
  than 'core show file version' make use of the file names, such as
  setting a debug level only on a specific file. As such, the act of
  registering source files with the Asterisk core still has use. The
  macro rename now reflects the new macro purpose.

* main/asterisk:
  - Refactor the file_version structure to reflect that it no longer
    tracks a version field.
  - Remove the "core show file version" CLI command. Without the file
    version, it is no longer useful.
  - Remove the ast_file_version_find function. The file version is no
    longer tracked.
  - Rename ast_register_file_version/ast_unregister_file_version to
    ast_register_file/ast_unregister_file, respectively.

* main/manager: Remove value from the Version key of the ModuleCheck
  Action. The actual key itself has not been removed, as doing so would
  absolutely constitute a backwards incompatible change. However, since
  the file version is no longer tracked, there is no need to attempt to
  include it in the Version key.

* UPGRADE: Add notes for:
  - Modification to the ModuleCheck AMI Action
  - Removal of the "core show file version" CLI command

Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e
2015-04-13 03:48:57 -04:00
Walter Doekes 49cbfa7de6 Fix typo's (retrieve, specified, address).
........

Merged revisions 430996 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 430998 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431000 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-23 15:13:08 +00:00
Walter Doekes 8b6ecc449c Fix printf problems with high ascii characters after r413586 (1.8).
In r413586 (1.8) various casts were added to silence gcc 4.10 warnings.
Those fixes included things like:

    -out += sprintf(out, "%%%02X", (unsigned char) *ptr);
    +out += sprintf(out, "%%%02X", (unsigned) *ptr);

That works for low ascii characters, but for the high range that yields
e.g. FFFFFFC3 when C3 is expected.

This changeset:
- fixes those casts to use the 'hh' unsigned char modifier instead
- consistently uses %02x instead of %2.2x (or other non-standard usage)
- adds a few 'h' modifiers in various places
- fixes a 'replcaes' typo
- dev/urandon typo (in 13+ patch)

Review: https://reviewboard.asterisk.org/r/4263/

ASTERISK-24619 #close
Reported by: Stefan27 (on IRC)
........

Merged revisions 429673 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 429674 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 429675 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429683 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-17 10:23:32 +00:00
Igor Goncharovskiy c866ced76b Add additional checks for NULL pointers to fix several crashes reported.
ASTERISK-24304 #close
Reported by: dhanapathy sathya
........

Merged revisions 426666 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 426667 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 426668 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426669 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-30 06:15:14 +00:00
Igor Goncharovskiy a770ca168d Fix loss of voice after second call drops (on a second line) in case using multiple lines on unistim phones. There is regression was introduced in r391379.
Reported by: Rustam Khankishyiev
(closes issue ASTERISK-23846)
........

Merged revisions 425667 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 425668 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 425669 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425677 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-16 06:22:07 +00:00
Richard Mudgett 0165c5f95a chan_pjsip: Fix deadlock when masquerading PJSIP channels.
Performing a directed call pickup resulted in a deadlock when PJSIP
channels were involved.

A masquerade needs to hold onto the channel locks while it swaps channel
information between the two channels involved in the masquerade.  With
PJSIP channels, the fixup routine needed to push a fixup task onto the
PJSIP channel's serializer.  Unfortunately, if the serializer was also
processing a task that needed to lock the channel, you get deadlock.

* Added a new control frame that is used to notify the channels that a
masquerade is about to start and when it has completed.

* Added the ability to query taskprocessors if the current thread is the
taskprocessor thread.

* Added the ability to suspend/unsuspend the PJSIP serializer thread so a
masquerade could fixup the PJSIP channel without using the serializer.

ASTERISK-24356 #close
Reported by: rmudgett

Review: https://reviewboard.asterisk.org/r/4034/
........

Merged revisions 424471 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 424472 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424473 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-03 17:47:42 +00:00
Mark Michelson dcf1ad14da Add module support level to ast_module_info structure. Print it in CLI "module show" .
ASTERISK-23919 #close
Reported by Malcolm Davenport

Review: https://reviewboard.asterisk.org/r/3802



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419592 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-25 16:47:17 +00:00
Matthew Jordan a2c912e997 media formats: re-architect handling of media for performance improvements
In the old times media formats were represented using a bit field. This was
fast but had a few limitations.
 1. Asterisk was limited in how many formats it could handle.
 2. Formats, being a bit field, could not include any attribute information.
    A format was strictly its type, e.g., "this is ulaw".
This was changed in Asterisk 10 (see
https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal for
notes on that work) which led to the creation of the ast_format structure.
This structure allowed Asterisk to handle attributes and bundle information
with a format.

Additionally, ast_format_cap was created to act as a container for multiple
formats that, together, formed the capability of some entity. Another
mechanism was added to allow logic to be registered which performed format
attribute negotiation. Everywhere throughout the codebase Asterisk was
changed to use this strategy.

Unfortunately, in software, there is no free lunch. These new capabilities
came at a cost.

Performance analysis and profiling showed that we spend an inordinate
amount of time comparing, copying, and generally manipulating formats and
their related structures. Basic prototyping has shown that a reasonably
large performance improvement could be made in this area. This patch is the
result of that project, which overhauled the media format architecture
and its usage in Asterisk to improve performance.

Generally, the new philosophy for handling formats is as follows:
 * The ast_format structure is reference counted. This removed a large amount
   of the memory allocations and copying that was done in prior versions.
 * In order to prevent race conditions while keeping things performant, the
   ast_format structure is immutable by convention and lock-free. Violate this
   tenet at your peril!
 * Because formats are reference counted, codecs are also reference counted.
   The Asterisk core generally provides built-in codecs and caches the
   ast_format structures created to represent them. Generally, to prevent
   inordinate amounts of module reference bumping, codecs and formats can be
   added at run-time but cannot be removed.
 * All compatibility with the bit field representation of codecs/formats has
   been moved to a compatibility API. The primary user of this representation
   is chan_iax2, which must continue to maintain its bit-field usage of formats
   for interoperability concerns.
 * When a format is negotiated with attributes, or when a format cannot be
   represented by one of the cached formats, a new format object is created or
   cloned from an existing format. That format may have the same codec
   underlying it, but is a different format than a version of the format with
   different attributes or without attributes.
 * While formats are reference counted objects, the reference count maintained
   on the format should be manipulated with care. Formats are generally cached
   and will persist for the lifetime of Asterisk and do not explicitly need
   to have their lifetime modified. An exception to this is when the user of a
   format does not know where the format came from *and* the user may outlive
   the provider of the format. This occurs, for example, when a format is read
   from a channel: the channel may have a format with attributes (hence,
   non-cached) and the user of the format may last longer than the channel (if
   the reference to the channel is released prior to the format's reference).

For more information on this work, see the API design notes:
  https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite

Finally, this work was the culmination of a large number of developer's
efforts. Extra thanks goes to Corey Farrell, who took on a large amount of the
work in the Asterisk core, chan_sip, and was an invaluable resource in peer
reviews throughout this project.

There were a substantial number of patches contributed during this work; the
following issues/patch names simply reflect some of the work (and will cause
the release scripts to give attribution to the individuals who work on them).

Reviews:
 https://reviewboard.asterisk.org/r/3814
 https://reviewboard.asterisk.org/r/3808
 https://reviewboard.asterisk.org/r/3805
 https://reviewboard.asterisk.org/r/3803
 https://reviewboard.asterisk.org/r/3801
 https://reviewboard.asterisk.org/r/3798
 https://reviewboard.asterisk.org/r/3800
 https://reviewboard.asterisk.org/r/3794
 https://reviewboard.asterisk.org/r/3793
 https://reviewboard.asterisk.org/r/3792
 https://reviewboard.asterisk.org/r/3791
 https://reviewboard.asterisk.org/r/3790
 https://reviewboard.asterisk.org/r/3789
 https://reviewboard.asterisk.org/r/3788
 https://reviewboard.asterisk.org/r/3787
 https://reviewboard.asterisk.org/r/3786
 https://reviewboard.asterisk.org/r/3784
 https://reviewboard.asterisk.org/r/3783
 https://reviewboard.asterisk.org/r/3778
 https://reviewboard.asterisk.org/r/3774
 https://reviewboard.asterisk.org/r/3775
 https://reviewboard.asterisk.org/r/3772
 https://reviewboard.asterisk.org/r/3761
 https://reviewboard.asterisk.org/r/3754
 https://reviewboard.asterisk.org/r/3753
 https://reviewboard.asterisk.org/r/3751
 https://reviewboard.asterisk.org/r/3750
 https://reviewboard.asterisk.org/r/3748
 https://reviewboard.asterisk.org/r/3747
 https://reviewboard.asterisk.org/r/3746
 https://reviewboard.asterisk.org/r/3742
 https://reviewboard.asterisk.org/r/3740
 https://reviewboard.asterisk.org/r/3739
 https://reviewboard.asterisk.org/r/3738
 https://reviewboard.asterisk.org/r/3737
 https://reviewboard.asterisk.org/r/3736
 https://reviewboard.asterisk.org/r/3734
 https://reviewboard.asterisk.org/r/3722
 https://reviewboard.asterisk.org/r/3713
 https://reviewboard.asterisk.org/r/3703
 https://reviewboard.asterisk.org/r/3689
 https://reviewboard.asterisk.org/r/3687
 https://reviewboard.asterisk.org/r/3674
 https://reviewboard.asterisk.org/r/3671
 https://reviewboard.asterisk.org/r/3667
 https://reviewboard.asterisk.org/r/3665
 https://reviewboard.asterisk.org/r/3625
 https://reviewboard.asterisk.org/r/3602
 https://reviewboard.asterisk.org/r/3519
 https://reviewboard.asterisk.org/r/3518
 https://reviewboard.asterisk.org/r/3516
 https://reviewboard.asterisk.org/r/3515
 https://reviewboard.asterisk.org/r/3512
 https://reviewboard.asterisk.org/r/3506
 https://reviewboard.asterisk.org/r/3413
 https://reviewboard.asterisk.org/r/3410
 https://reviewboard.asterisk.org/r/3387
 https://reviewboard.asterisk.org/r/3388
 https://reviewboard.asterisk.org/r/3389
 https://reviewboard.asterisk.org/r/3390
 https://reviewboard.asterisk.org/r/3321
 https://reviewboard.asterisk.org/r/3320
 https://reviewboard.asterisk.org/r/3319
 https://reviewboard.asterisk.org/r/3318
 https://reviewboard.asterisk.org/r/3266
 https://reviewboard.asterisk.org/r/3265
 https://reviewboard.asterisk.org/r/3234
 https://reviewboard.asterisk.org/r/3178

ASTERISK-23114 #close
Reported by: mjordan
  media_formats_translation_core.diff uploaded by kharwell (License 6464)
  rb3506.diff uploaded by mjordan (License 6283)
  media_format_app_file.diff uploaded by kharwell (License 6464) 
  misc-2.diff uploaded by file (License 5000)
  chan_mild-3.diff uploaded by file (License 5000) 
  chan_obscure.diff uploaded by file (License 5000) 
  jingle.diff uploaded by file (License 5000) 
  funcs.diff uploaded by file (License 5000) 
  formats.diff uploaded by file (License 5000) 
  core.diff uploaded by file (License 5000) 
  bridges.diff uploaded by file (License 5000) 
  mf-codecs-2.diff uploaded by file (License 5000) 
  mf-app_fax.diff uploaded by file (License 5000) 
  mf-apps-3.diff uploaded by file (License 5000) 
  media-formats-3.diff uploaded by file (License 5000) 

ASTERISK-23715
  rb3713.patch uploaded by coreyfarrell (License 5909)
  rb3689.patch uploaded by mjordan (License 6283)
  
ASTERISK-23957
  rb3722.patch uploaded by mjordan (License 6283) 
  mf-attributes-3.diff uploaded by file (License 5000) 

ASTERISK-23958
Tested by: jrose
  rb3822.patch uploaded by coreyfarrell (License 5909) 
  rb3800.patch uploaded by jrose (License 6182)
  chan_sip.diff uploaded by mjordan (License 6283) 
  rb3747.patch uploaded by jrose (License 6182)

ASTERISK-23959 #close
Tested by: sgriepentrog, mjordan, coreyfarrell
  sip_cleanup.diff uploaded by opticron (License 6273)
  chan_sip_caps.diff uploaded by mjordan (License 6283) 
  rb3751.patch uploaded by coreyfarrell (License 5909) 
  chan_sip-3.diff uploaded by file (License 5000) 

ASTERISK-23960 #close
Tested by: opticron
  direct_media.diff uploaded by opticron (License 6273) 
  pjsip-direct-media.diff uploaded by file (License 5000) 
  format_cap_remove.diff uploaded by opticron (License 6273) 
  media_format_fixes.diff uploaded by opticron (License 6273) 
  chan_pjsip-2.diff uploaded by file (License 5000) 

ASTERISK-23966 #close
Tested by: rmudgett
  rb3803.patch uploaded by rmudgetti (License 5621)
  chan_dahdi.diff uploaded by file (License 5000) 
  
ASTERISK-24064 #close
Tested by: coreyfarrell, mjordan, opticron, file, rmudgett, sgriepentrog, jrose
  rb3814.patch uploaded by rmudgett (License 5621) 
  moh_cleanup.diff uploaded by opticron (License 6273) 
  bridge_leak.diff uploaded by opticron (License 6273) 
  translate.diff uploaded by file (License 5000) 
  rb3795.patch uploaded by rmudgett (License 5621) 
  tls_fix.diff uploaded by mjordan (License 6283) 
  fax-mf-fix-2.diff uploaded by file (License 5000) 
  rtp_transfer_stuff uploaded by mjordan (License 6283) 
  rb3787.patch uploaded by rmudgett (License 5621) 
  media-formats-explicit-translate-format-3.diff uploaded by file (License 5000) 
  format_cache_case_fix.diff uploaded by opticron (License 6273) 
  rb3774.patch uploaded by rmudgett (License 5621) 
  rb3775.patch uploaded by rmudgett (License 5621) 
  rtp_engine_fix.diff uploaded by opticron (License 6273) 
  rtp_crash_fix.diff uploaded by opticron (License 6273) 
  rb3753.patch uploaded by mjordan (License 6283) 
  rb3750.patch uploaded by mjordan (License 6283) 
  rb3748.patch uploaded by rmudgett (License 5621) 
  media_format_fixes.diff uploaded by opticron (License 6273) 
  rb3740.patch uploaded by mjordan (License 6283) 
  rb3739.patch uploaded by mjordan (License 6283) 
  rb3734.patch uploaded by mjordan (License 6283) 
  rb3689.patch uploaded by mjordan (License 6283) 
  rb3674.patch uploaded by coreyfarrell (License 5909) 
  rb3671.patch uploaded by coreyfarrell (License 5909) 
  rb3667.patch uploaded by coreyfarrell (License 5909) 
  rb3665.patch uploaded by mjordan (License 6283) 
  rb3625.patch uploaded by coreyfarrell (License 5909) 
  rb3602.patch uploaded by coreyfarrell (License 5909) 
  format_compatibility-2.diff uploaded by file (License 5000) 
  core.diff uploaded by file (License 5000) 
  


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-20 22:06:33 +00:00
Matthew Jordan fb5690ce4b Logger/CLI/etc.: Fix some aesthetic issues; reduce chatty verbose messages
This patch addresses some aesthetic issues in Asterisk. These are all just
minor tweaks to improve the look of the CLI when used in a variety of
settings. Specifically:
 * A number of chatty verbose messages were removed or demoted to DEBUG
   messages. Verbose messages with a verbosity level of 5 or higher were -
   if kept as verbose messages - demoted to level 4. Several messages
   that were emitted at verbose level 3 were demoted to 4, as announcement
   of dialplan applications being executed occur at level 3 (and so the
   effects of those applications should generally be less).
 * Some verbose messages that only appear when their respective 'debug'
   options are enabled were bumped up to always be displayed.
 * Prefix/timestamping of verbose messages were moved to the verboser
   handlers. This was done to prevent duplication of prefixes when the
   timestamp option (-T) is used with the CLI.
 * Verbose magic is removed from messages before being emitted to
   non-verboser handlers. This prevents the magic in multi-line verbose
   messages (such as SIP debug traces or the output of DumpChan) from
   being written to files.
 * _Slightly_ better support for the "light background" option (-W) was
   added. This includes using ast_term_quit in the output of XML
   documentation help, as well as changing the "Asterisk Ready" prompt to
   bright green on the default background (which stands a better chance of
   being displayed properly than bright white).

Review: https://reviewboard.asterisk.org/r/3547/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-28 22:54:12 +00:00
Walter Doekes e5194c91fc chan_unistim: Unlock mutex in rare OOM condition.
#ASTERISK-23792 #close
Reported by: Peter Whisker

Review: https://reviewboard.asterisk.org/r/3567/
........

Merged revisions 414677 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 414678 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414679 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-28 09:43:53 +00:00
Kinsey Moore abd3e4040b Allow Asterisk to compile under GCC 4.10
This resolves a large number of compiler warnings from GCC 4.10 which
cause the build to fail under dev mode. The vast majority are
signed/unsigned mismatches in printf-style format strings.
........

Merged revisions 413586 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 413587 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 413588 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-09 22:49:26 +00:00
Igor Goncharovskiy d3433771c9 Introducing changes proposed to chan_unistim driver:
1) Added the unistim.conf variable dtmf_duration which can select the DTMF playback duration from 0ms to 150ms (0 is off and is the new default)
2) Enabled the transmission of month names, which are sent with the date and changed the dateformat variable to accept the values 0-3 as per the UNISTIM standard (2 & 3 match the previous 1 & 2 formats).
3) Enabled the "Mute" packet so muting microphone works as expected and microphone muted for all calls while LED light on
4) Changed Duree to Timer on i2004 display

(closes issue ASTERISK-23592)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413048 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-04-28 07:43:33 +00:00
Igor Goncharovskiy cb6d928a39 Fix wrong dialtone. The "modulation" should not be referenced for tone+tone as it refers to the on-off characteristic - this often resulted in a single tone rather than the multitone as in the UK.
........

Merged revisions 412712 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 412713 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412714 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-04-21 08:36:18 +00:00
Scott Griepentrog 80ef9a21b9 uniqueid: channel linkedid, ami, ari object creation with id's
Much needed was a way to assign id to objects on creation, and
much change was necessary to accomplish it.  Channel uniqueids
and linkedids are split into separate string and creation time
components without breaking linkedid propgation.  This allowed
the uniqueid to be specified by the user interface - and those
values are now carried through to channel creation, adding the
assignedids value to every function in the chain including the
channel drivers. For local channels, the second channel can be
specified or left to default to a ;2 suffix of first.  In ARI,
bridge, playback, and snoop objects can also be created with a
specified uniqueid.

Along the way, the args order to allocating channels was fixed
in chan_mgcp and chan_gtalk, and linkedid is no longer lost as
masquerade occurs.

(closes issue ASTERISK-23120)
Review: https://reviewboard.asterisk.org/r/3191/
........

Merged revisions 410157 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-07 15:47:55 +00:00
Igor Goncharovskiy fe73de1302 Add update_peer function to unistim_rtp_glue, improve other unistim_rtp_glue 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
2014-03-05 06:17:03 +00:00
Igor Goncharovskiy 9397e7af2a Implement functions handling keypress, display icons and text for i2004 KEM support.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408932 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-26 08:57:14 +00:00
Richard Mudgett e4803bbd9e Voicemail: Remove mailbox identifier format (box@context) assumptions in the system.
This change is in preparation for external MWI support.

Removed code from the system for normal mailbox handling that appends
@default to the mailbox identifier if it does not have a context.  The
only exception is the legacy hasvoicemail users.conf option.  The legacy
option will only work for app_voicemail mailboxes.  The system cannot make
any assumptions about the format of the mailbox identifer used by
app_voicemail.

chan_sip and chan_dahdi/sig_pri had the most changes because they both
tried to interpret the mailbox identifier.  chan_sip just stored and
compared the two components.  chan_dahdi actually used the box
information.

The ISDN MWI support configuration options had to be reworked because
chan_dahdi was parsing the box@context format to get the box number.  As a
result the mwi_vm_boxes chan_dahdi.conf option was added and is documented
in the chan_dahdi.conf.sample file.

Review: https://reviewboard.asterisk.org/r/3072/
........

Merged revisions 404348 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404350 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-19 16:52:43 +00:00
Kevin Harwell 28c0cb28d0 channel locking: Add locking for channel snapshot creation
Original commit message by mmichelson (asterisk 12 r403311):

"This adds channel locks around calls to create channel snapshots as well
as other functions which operate on a channel and then end up
creating a channel snapshot. Functions that expect the channel to be
locked prior to being called have had their documentation updated to
indicate such."

The above was initially committed and then reverted at r403398.  The problem
was found to be in core_local.c in the publish_local_bridge_message function.
The ast_unreal_lock_all function locks and adds a reference to the returned
channels and while they were being unlocked they were not being unreffed when
no longer needed.  Fixed by unreffing the channels.

Also in bridge.c a lock was obtained on "other->chan", but then an attempt was
made to unlock "other" and not the previously locked channel.  Fixed by
unlocking "other->chan"

(closes issue ASTERISK-22709)
Reported by: John Bigelow
........

Merged revisions 404237 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404260 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-18 20:33:37 +00:00
Joshua Colp e2630fcd51 channels: Return allocated channels locked.
This change makes ast_channel_alloc return allocated channels
locked. By doing so no other thread can acquire, lock, and manipulate
the channel before it is completely set up.

(closes issue AST-1256)

Review: https://reviewboard.asterisk.org/r/3067/
........

Merged revisions 404204 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404210 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-18 19:28:05 +00:00
David M. Lee 1212906351 Reverting r403311. It's causing ARI tests to hang.
........

Merged revisions 403398 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-05 22:10:20 +00:00
Mark Michelson 8e8b329e14 Add channel locking for channel snapshot creation.
This adds channel locks around calls to create channel snapshots as well
as other functions which operate on a channel and then end up
creating a channel snapshot. Functions that expect the channel to be
locked prior to being called have had their documentation updated to
indicate such.
........

Merged revisions 403311 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-03 17:07:29 +00:00
Mark Michelson ee21eee7e0 Cache string values of formats on ast_format_cap() to save processing.
Channel snapshots have string representations of the channel's native formats.
Prior to this change, the format strings were re-created on ever channel snapshot
creation. Since channel native formats rarely change, this was very wasteful.
Now, string representations of formats may optionally be stored on the ast_format_cap
for cases where string representations may be requested frequently. When formats
are altered, the string cache is marked as invalid. When strings are requested, the
cache validity is checked. If the cache is valid, then the cached strings are copied.
If the cache is invalid, then the string cache is rebuilt and copied, and the cache
is marked as being valid again.

Review: https://reviewboard.asterisk.org/r/2879
........

Merged revisions 400356 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400363 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-03 14:58:16 +00:00
Joshua Colp c1235f2639 Reduce channel snapshot creation and publishing by up to 50%.
This change introduces the ability to stage channel snapshot
creation and publishing by suppressing the implicit creation
and publishing that some functions have. Once all operations
are executed the staging is marked as done and a single snapshot
is created and published.

Review: https://reviewboard.asterisk.org/r/2889/
........

Merged revisions 400265 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400266 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-02 16:23:34 +00:00
Kinsey Moore 59753b1ea1 Strip down the old event system
This removes unused code, event types, IE pltypes, and event IE types
where possible and makes several functions private that were once
public. This includes a renumbering of the remaining event and IE types
which breaks binary compatibility with previous versions. The last
remaining consumers of the old event system (or parts thereof) are
main/security_events.c, res/res_security_log.c, tests/test_cel.c,
tests/test_event.c, main/cel.c, and the CEL backends.

Review: https://reviewboard.asterisk.org/r/2703/
(closes issue ASTERISK-22139)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-17 14:39:27 +00:00
Igor Goncharovskiy 70309f24c2 - Fix different issues with call transfer cancel. In case 3rd party busy or congestion call was not returned.
- Fix displaying soft button 'Redial' in case of no redial number exists
........

Merged revisions 396377 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-08 07:05:54 +00:00
Matthew Jordan 38236e54a8 Remove dead code from features.c; refactor pickup code into pickup.c
This patch does the following:
 * It moves the pickup code out of features.c and into pickup.c
 * It removes the vast majority of dead code out of features.c. In particular,
   this includes the parking code.

(issue ASTERISK-22134)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-02 02:32:44 +00:00
David M. Lee e1b959ccbb Split caching out from the stasis_caching_topic.
In working with res_stasis, I discovered a significant limitation to
the current structure of stasis_caching_topics: you cannot subscribe
to cache updates for a single channel/bridge/endpoint/etc.

To address this, this patch splits the cache away from the
stasis_caching_topic, making it a first class object. The stasis_cache
object is shared amongst individual stasis_caching_topics that are
created per channel/endpoint/etc. These are still forwarded to global
whatever_all_cached topics, so their use from most of the code does
not change.

In making these changes, I noticed that we frequently used a similar
pattern for bridges, endpoints and channels:

     single_topic  ---------------->  all_topic
           ^
           |
     single_topic_cached  ----+---->  all_topic_cached
                              |
                              +---->  cache

This pattern was extracted as the 'Stasis Caching Pattern', defined in
stasis_caching_pattern.h. This avoids a lot of duplicate code between
the different domain objects.

Since the cache is now disassociated from its upstream caching topics,
this also necessitated a change to how the 'guaranteed' flag worked
for retrieving from a cache. The code for handling the caching
guarantee was extracted into a 'stasis_topic_wait' function, which
works for any stasis_topic.

(closes issue ASTERISK-22002)
Review: https://reviewboard.asterisk.org/r/2672/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395954 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-01 13:49:34 +00:00
Matthew Jordan cafc115896 A great big renaming patch
This patch renames the bridging* files to bridge*. This may seem pedantic
and silly, but it fits better in line with current Asterisk naming conventions:
* channel is not "channeling"
* monitor is not "monitoring"
etc.

A bridge is an object. It is a first class citizen in Asterisk. "Bridging" is
the act of using a bridge on a set of channels - and the API that fulfills that
role is more than just the action.

(closes issue ASTERISK-22130)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-25 04:06:32 +00:00
Matthew Jordan d0a55fa52d Refactor RTCP events over to Stasis; associate with channels
This patch does the following:

* It merges Jaco Kroon's patch from ASTERISK-20754, which provides channel
  information in the RTCP events. Because Stasis provides a cache, Jaco's
  patch was modified to pass the channel uniqueid to the RTP layer as
  opposed to a pointer to the channel. This has the following benefits:
  (1) It keeps the RTP engine 'clean' of references back to channels
  (2) It prevents circular dependencies and other potential ref counting issues
* The RTP engine now allows any RTP implementation to raise RTCP messages.
  Potentially, other implementations (such as res_rtp_multicast) could also
  raise RTCP information. The engine provides structs to represent RTCP headers
  and RTCP SR/RR reports.
* Some general refactoring in res_rtp_asterisk was done to try and tame the
  RTCP code. It isn't perfect - that's *way* beyond the scope of this work -
  but it does feel marginally better.
* A few random bugs were fixed in the RTCP statistics. (Example: performing an
  assignment of a = a is probably not correct)
* We now raise RTCP events for each SR/RR sent/received. Previously we wouldn't
  raise an event when we sent a RR report.

Note that this work will be of use to others who want to monitor call quality
or build modules that report call quality statistics. Since the events are now
moving across the Stasis message bus, this is far easier to accomplish. It is
also a first step (though by no means the last step) towards getting Olle's
pinefrog work incorporated.

Again: note that the patch by Jaco Kroon was modified slightly for this work;
however, he did all of the hard work in finding the right places to set the
channel in the RTP engine across the channel drivers. Much thanks goes to Jaco
for his hard work here.

Review: https://reviewboard.asterisk.org/r/2603/

(closes issue ASTERISK-20574)
Reported by: Jaco Kroon
patches:
  asterisk-rtcp-channel.patch uploaded by jkroon (License 5671)

(closes issue ASTERISK-21471)
Reported by: Matt Jordan



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393740 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-05 17:33:33 +00:00
Igor Goncharovskiy f7624718f8 Fix issue with inability to cancell call transfer made by on-sceen menus.
Reported by: Igor Olhovskiy
........

Merged revisions 393395 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393396 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-02 10:16:27 +00:00
Mark Michelson e3a89a0a18 Change chan_unistim to use core transfer API.
Review: https://reviewboard.asterisk.org/r/2553

(closes issue ASTERISK-21527)
Reported by Matt Jordan



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-21 18:05:56 +00:00
Matthew Jordan 6258bbe7bd Update Asterisk's CDRs for the new bridging framework
This patch is the initial push to update Asterisk's CDR engine for the new
bridging framework. This patch guts the existing CDR engine and builds the new
on top of messages coming across Stasis. As changes in channel state and bridge
state are detected, CDRs are built and dispatched accordingly. This
fundamentally changes CDRs in a few ways.
(1) CDRs are now *very* reflective of the actual state of channels and bridges.
    This means CDRs track well with what an actual channel is doing - which
    is useful in transfer scenarios (which were previously difficult to pin
    down). It does, however, mean that CDRs cannot be 'fooled'. Previous
    behavior in Asterisk allowed for CDR applications, channels, and other
    properties to be spoofed in parts of the code - this no longer works.
(2) CDRs have defined behavior in multi-party scenarios. This behavior will not
    be what everyone wants, but it is a defined behavior and as such, it is
    predictable.
(3) The CDR manipulation functions and applications have been overhauled. Major
    changes have been made to ResetCDR and ForkCDR in particular. Many of the
    options for these two applications no longer made any sense with the new
    framework and the (slightly) more immutable nature of CDRs.

There are a plethora of other changes. For a full description of CDR behavior,
see the CDR specification on the Asterisk wiki.

(closes issue ASTERISK-21196)

Review: https://reviewboard.asterisk.org/r/2486/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391947 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-17 03:00:38 +00:00
Igor Goncharovskiy 2053fc3159 Fix issue with no sound in both way in case of previous call to chan_unistim phone was canceled.
(related to ASTERISK-20183)
........

Merged revisions 391379 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391380 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-11 10:24:04 +00:00
Mark Michelson 2dc8a06006 Refactor the features configuration scheme.
Features configuration is handled in its own API in
features_config.h and features_config.c. This way, features
configuration is accessible to anything that needs it.

In addition, features configuration has been altered to
be more channel-oriented. Most callers of features API
code will be supplying a channel so that the individual
channel's settings will be acquired rather than the global
setting.

Missing from this commit is XML documentation for the
features configuration. That will be handled in a separate
commit.

Review: https://reviewboard.asterisk.org/r/2578/

(issue ASTERISK-21542)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390751 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-06 21:40:35 +00:00
Jason Parker 154fbf8cae Split Hold event into Hold/Unhold, and move it into core.
(closes issue ASTERISK-21487)
Review: https://reviewboard.asterisk.org/r/2565/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389746 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-24 21:21:25 +00:00
Matthew Jordan 06be8463b6 Migrate a large number of AMI events over to Stasis-Core
This patch moves a number of AMI events over to the Stasis-Core message bus.
This includes:
 * ChanSpyStart/Stop
 * MonitorStart/Stop
 * MusicOnHoldStart/Stop
 * FullyBooted/Reload
 * All Voicemail/MWI related events

In addition, it adds some Stasis-Core and AMI support for generic AMI messages,
refactors the message router in AMI to use a single router with topic
forwarding for the topics that AMI cares about, and refactors MWI message
types and topics to be more name compliant.

Review: https://reviewboard.asterisk.org/r/2532

(closes issue ASTERISK-21462)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389733 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-24 20:44:07 +00:00
Igor Goncharovskiy 1fb6f365ec Fix several problems caused by multiple line usage with i2004 phones.
Reported by: Daniel Bohling, MihaiMircea

(closes issue ASTERISK-21061)
(closes issue ASTERISK-21120)
........

Merged revisions 389661 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389663 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-24 10:23:48 +00:00
Richard Mudgett 3d63833bd6 Merge in the bridge_construction branch to make the system use the Bridging API.
Breaks many things until they can be reworked.  A partial list:
chan_agent
chan_dahdi, chan_misdn, chan_iax2 native bridging
app_queue
COLP updates
DTMF attended transfers
Protocol attended transfers


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-21 18:00:22 +00:00
Richard Mudgett 5fec9b8d1f Remove some unnecessary calls to ast_bridged_channel() in chan_unistim.c
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387185 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-01 20:01:43 +00:00
Kinsey Moore 1a2a4578d2 Convert MWI state message type to the new stasis naming convention
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384219 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-27 22:42:06 +00:00
Kinsey Moore 99aa02d17f Transition MWI to Stasis-core
Remove MWI's dependency on the event system by moving it to
Stasis-core. This also introduces forwarding topic pools in Stasis-core
which aggregate many dynamically allocated topics into a single primary
topic.

Review: https://reviewboard.asterisk.org/r/2368/
(closes issue ASTERISK-21097)
Patch-by: Kinsey Moore


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-16 15:45:58 +00:00
Igor Goncharovskiy ef64b29f8b Fix core dump on CLI usage
Fix issue with 'unistim show info' CLI command when device connected not configured
........

Merged revisions 382827 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382828 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-12 08:55:14 +00:00
Igor Goncharovskiy 469ca1c71d Fix several unreleased mutex locks that cause problem with processing calls
Reported by: Daniel Bohling
Tested by: Daniel Bohling

(Closes issue ASTERISK-21119)
........

Merged revisions 382409 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 382410 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382411 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-05 03:53:44 +00:00