Commit Graph

18 Commits

Author SHA1 Message Date
Corey Farrell
90a11c4ae7 chan_sip deprecation.
This officially deprecates chan_sip in Asterisk 17+.  A warning is
printed upon startup or module load to tell users that they should
consider migrating.  chan_sip is still built by default but the default
modules.conf skips loading it at startup.

Very important to note we are not scheduling a time where chan_sip will
be removed.  The goal of this change is to accurately inform end users
of the current state of chan_sip and encourage movement to the fully
supported chan_pjsip.

Change-Id: Icebd8848f63feab94ef882d36b2e99d73155af93
2018-10-25 08:57:16 -04:00
Sean Bright
fd0ca1c3f9 Remove as much trailing whitespace as possible.
Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0
2017-12-22 09:23:22 -05:00
Richard Mudgett
8536a09b86 security-events: Fix SuccessfulAuth using_password declaration.
The SuccessfulAuth using_password field was declared as a pointer to a
uint32_t when the field was later read as a uint32_t value.  This resulted
in unnecessary casts and a non-portable field value reinterpret in
main/security_events.c:add_json_object().  i.e., It would work on a 32 bit
architecture but not on a 64 bit big endian architecture.

Change-Id: Ia08bc797613a62f07e5473425f9ccd8d77c80935
2017-12-04 17:21:27 -06:00
Corey Farrell
a6e5bae3ef Remove ASTERISK_REGISTER_FILE.
ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes
all traces of it.

Previously exported symbols removed:
* __ast_register_file
* __ast_unregister_file
* ast_complete_source_filename

This also removes the mtx_prof static variable that was declared when
MTX_PROFILE was enabled.  This variable was only used in lock.c so it
is now initialized in that file only.

ASTERISK-26480 #close

Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966
2016-10-27 09:53:55 -04:00
Kevin Harwell
43bdddfc26 chan_sip.c: wrong peer searched in sip_report_security_event
In chan_sip, after handling an incoming invite a security event is raised
describing authorization (success, failure, etc...). However, it was doing
a lookup of the peer by extension. This is fine for register messages, but
in the case of an invite it may search and find the wrong peer, or a non
existent one (for instance, in the case of call pickup). Also, if the peers
are configured through realtime this may cause an unnecessary database lookup
when caching is enabled.

This patch makes it so that sip_report_security_event searches by IP address
when looking for a peer instead of by extension after an invite is processed.

ASTERISK-25320 #close

Change-Id: I9b3f11549efb475b6561c64f0e6da1a481d98bc4
2015-08-13 15:01:58 -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
Richard Mudgett
a7c9f4c668 ast_str: Fix improper member access to struct ast_str members.
Accessing members of struct ast_str outside of the string manipulation API
routines is invalid since struct ast_str is supposed to be treated as
opaque.

Review: https://reviewboard.asterisk.org/r/4194/
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Merged revisions 428244 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 428245 from http://svn.asterisk.org/svn/asterisk/branches/12
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Merged revisions 428246 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428255 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-11-19 17:22:29 +00:00
Matthew Jordan
91f7b66183 chan_sip: Mark chan_sip and its files as extended support
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Merged revisions 420562 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420563 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-08 17:53:39 +00:00
Matthew Jordan
4f30c7e91f main/astobj2: Make REF_DEBUG a menuselect item; improve REF_DEBUG output
This patch does the following:
(1) It makes REF_DEBUG a meneselect item. Enabling REF_DEBUG now enables
    REF_DEBUG globally throughout Asterisk.
(2) The ref debug log file is now created in the AST_LOG_DIR directory.
    Every run will now blow away the previous run (as large ref files
    sometimes caused issues). We now also no longer open/close the file
    on each write, instead relying on fflush to make sure data gets written
    to the file (in case the ao2 call being performed is about to cause a
    crash)
(3) It goes with a comma delineated format for the ref debug file. This
    makes parsing much easier. This also now includes the thread ID of the
    thread that caused ref change.
(4) A new python script instead for refcounting has been added in the
    contrib/scripts folder.
(5) The old refcounter implementation in utils/ has been removed.

Review: https://reviewboard.asterisk.org/r/3377/
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Merged revisions 412114 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 412115 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 412153 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412154 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-04-11 02:59:19 +00:00
Jonathan Rose
b90bba7a30 Stasis: Update security events to use Stasis
Also moves ACL messages to the security topic and gets rid of the
ACL topic

(closes issue ASTERISK-21103)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2496/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388975 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-17 17:36:10 +00:00
Matthew Jordan
0ffce56f1b AST-2013-003: Prevent username disclosure in SIP channel driver
When authenticating a SIP request with alwaysauthreject enabled, allowguest
disabled, and autocreatepeer disabled, Asterisk discloses whether a user
exists for INVITE, SUBSCRIBE, and REGISTER transactions in multiple ways. The
information is disclosed when:
 * A "407 Proxy Authentication Required" response is sent instead of a
   "401 Unauthorized" response
 * The presence or absence of additional tags occurs at the end of "403
   Forbidden" (such as "(Bad Auth)")
 * A "401 Unauthorized" response is sent instead of "403 Forbidden" response
   after a retransmission
 * Retransmission are sent when a matching peer did not exist, but not when a
   matching peer did exist.

This patch resolves these various vectors by ensuring that the responses sent
in all scenarios is the same, regardless of the presence of a matching peer.

This issue was reported by Walter Doekes, OSSO B.V. A substantial portion of
the testing and the solution to this problem was done by Walter as well - a
huge thanks to his tireless efforts in finding all the ways in which this
setting didn't work, providing automated tests, and working with Kinsey on
getting this fixed.

(closes issue ASTERISK-21013)
Reported by: wdoekes
Tested by: wdoekes, kmoore
patches:
  AST-2013-003-1.8 uploaded by kmoore, wdoekes (License 6273, 5674)
  AST-2013-003-10 uploaded by kmoore, wdoekes (License 6273, 5674)
  AST-2013-003-11 uploaded by kmoore, wdoekes (License 6273, 5674)
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Merged revisions 384003 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-27 15:27:31 +00:00
Kinsey Moore
81fa307af7 Fix some more REF_DEBUG-related build errors
When sip_ref_peer and sip_unref_peer were exported to be usable in
channels/sip/security_events.c, modifications to those functions when
building under REF_DEBUG were not taken into account. This change
moves the necessary defines into sip.h to make them accessible to
other parts of chan_sip that need them.
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Merged revisions 381282 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381285 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-12 20:18:21 +00:00
Kinsey Moore
e5210366e4 Clean up chan_sip
This clean up was broken out from
https://reviewboard.asterisk.org/r/1976/ and addresses the following:
 - struct sip_refer converted to use the stringfields API.
 - sip_{refer|notify}_allocate -> sip_{notify|refer}_alloc to match
   other *alloc functions.
 - Replace get_msg_text, get_msg_text2 and get_pidf_body -> No, not
   get_pidf_msg_text_body3 but get_content, to match add_content.
 - get_body doesn't get the request body, renamed to get_content_line.
 - get_body_by_line doesn't get the body line, and is just a simple if
   test. Moved code inline and removed function.
 - Remove camelCase in struct sip_peer peer state variables,
   onHold -> onhold, inUse -> inuse, inRinging -> ringing.
 - Remove camelCase in struct sip_request rlPart1 -> rlpart1,
   rlPart2 -> rlpart2.
 - Rename instances of pvt->randdata to pvt->nonce because that is what
   it is, no need to update struct sip_pvt because _it already has a
   nonce field_.
 - Removed struct sip_pvt randdata stringfield.
 - Remove useless (and inconsistent) 'header' suffix on variables in
   handle_request_subscribe.
 - Use ast_strdupa on Event header in handle_request_subscribe to avoid
   overly complicated strncmp calls to find the event package.
 - Move get_destination check in handle_request_subscribe to avoid
   duplicate checking for packages that don't need it.
 - Move extension state callback management in handle_request_subscribe
   to avoid duplicate checking for packages that don't need it.
 - Remove duplicate append_date prototype.
 - Rename append_date -> add_date to match other add_xxx functions.
 - Added add_expires helper function, removed code that manually added
   expires header.
 - Remove _header suffix on add_diversion_header (no other header adding
   functions have this).
 - Don't pass req->debug to request handle_request_XXXXX handlers if req
   is also being passed.
 - Don't pass req->ignore to check_auth as req is already being passed.
 - Don't create a subscription in handle_request_subscribe if
   p->expiry == 0.
 - Don't walk of the back of referred_by_name when splitting string in
   get_refer_info
 - Remove duplicate check for no dialog in handle_incoming when
   sipmethod == SIP_REFER, handle_request_refer checks for that.

Review: https://reviewboard.asterisk.org/r/1993/
Patch-by: gareth


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370636 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-31 19:10:41 +00:00
Joshua Colp
e938737570 Add support for SIP over WebSocket.
This allows SIP traffic to be exchanged over a WebSocket connection which is useful for rtcweb.

Review: https://reviewboard.asterisk.org/r/2008


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-16 12:35:04 +00:00
Kevin P. Fleming
166b4e2b30 Multiple revisions 369001-369002
........
  r369001 | kpfleming | 2012-06-15 10:56:08 -0500 (Fri, 15 Jun 2012) | 11 lines
  
  Add support-level indications to many more source files.
  
  Since we now have tools that scan through the source tree looking for files
  with specific support levels, we need to ensure that every file that is
  a component of a 'core' or 'extended' module (or the main Asterisk binary)
  is explicitly marked with its support level. This patch adds support-level
  indications to many more source files in tree, but avoids adding them to
  third-party libraries that are included in the tree and to source files
  that don't end up involved in Asterisk itself.
........
  r369002 | kpfleming | 2012-06-15 10:57:14 -0500 (Fri, 15 Jun 2012) | 3 lines
  
  Add a script to enable finding source files without support-levels defined.
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Merged revisions 369001-369002 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 369005 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369013 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-15 16:20:16 +00:00
Michael L. Young
abf40d9b28 Add IPv6 address support to security events framework.
The current Security Events Framework API only supports IPv4 when it comes to
generating security events.  This patch does the following:

* Changes the Security Events Framework API to support IPV6 and updates
  the components that use this API.

* Eliminates an error message that was being generated since the current
  implementation was treating an IPv6 socket address as if it was IPv4.

* Some copyright dates were updated on files touched by this patch.

(closes issue ASTERISK-19447) 
Reported by: Michael L. Young 
Tested by: Michael L. Young 
Patches: 
  security_events_ipv6v3.diff uploaded by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/1777/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-16 21:20:50 +00:00
Mark Michelson
1bef7695ce Add a security event for the case where fake authentication challenge is sent.
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Merged revisions 357318 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357319 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-28 18:52:13 +00:00
Jonathan Rose
5982bdcb7c Merged revisions 337595,337597 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

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  r337595 | jrose | 2011-09-22 10:35:50 -0500 (Thu, 22 Sep 2011) | 12 lines
  
  Generate Security events in chan_sip using new Security Events Framework
  
  Security Events Framework was added in 1.8 and support was added for AMI to generate
  events at that time. This patch adds support for chan_sip to generate security events.
  
  (closes issue ASTERISK-18264)
  Reported by: Michael L. Young
  Patches:
       security_events_chan_sip_v4.patch (license #5026) by Michael L. Young
  Review: https://reviewboard.asterisk.org/r/1362/
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  r337597 | jrose | 2011-09-22 10:47:05 -0500 (Thu, 22 Sep 2011) | 10 lines
  
  Forgot to svn add new files to r337595
  
  Part of Generating security events for chan_sip
  
  (issue ASTERISK-18264)
  Reported by: Michael L. Young
  Patches:
      security_events_chan_sip_v4.patch (License #5026) by Michael L. Young
  Reviewboard: https://reviewboard.asterisk.org/r/1362/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337600 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-22 16:35:20 +00:00