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20 authors
  1. Feb 15, 2015
  3. Feb 14, 2015
  5. Feb 13, 2015
    • Richard Mudgett's avatar
      res_pjsip_session: Fix double re-INVITE collision crash. · f00ebf0a
      Richard Mudgett authored 
      A multi-asterisk box setup with direct media enabled would occasionally
crash when two re-INVITE collisions on a call leg happen in a row.

The re-INVITE logic only had one timer struct to defer the re-INVITE.
When the second collision happens the timer struct is overwritten and put
into the timer heap again.  Resources for the first timer are leaked and
the heap has two positions occupied by the same timer struct.  Now the
heap ordering is potentially corrupted, the timer will fire twice, and any
resources allocated for the second timer will be released twice.

* The solution is to put the collided re-INVITE into the delayed requests
queue with all the other delayed requests and cherry pick the next request
that can come off the queue when an event happens.

* Changed to put delayed BYE requests at the head of the delayed queue.
There is no sense in processing delayed UPDATEs and re-INVITEs when a BYE
has been requested.

* Made the start of a BYE request flush the delayed requests queue to
prevent a delayed request from overlapping the BYE transaction.  I saw a
few cases where a delayed re-INVITE got started after the BYE transaction
started.

* Changed the delayed_request struct to use an enum instead of a string
for the request method.  Cherry picking the queue is easier with an enum
than string comparisons and the compiler can warn if a switch statement
does not cover all defined enum values.

* Improved the debug output to give more information.  It helps to know
which channel is involved with an endpoint.  Trunks can have many channels
associated with the endpoint at the same time.

ASTERISK-24727 #close
Reported by: Mark Michelson

Review: https://reviewboard.asterisk.org/r/4414/
........

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      f00ebf0a
  7. Feb 12, 2015
    • Matthew Jordan's avatar
      ARI/PJSIP: Add the ability to redirect (transfer) a channel in a Stasis app · 29f66b04
      Matthew Jordan authored 
      This patch adds a new feature to ARI to redirect a channel to another server,
and fixes a few bugs in PJSIP's handling of the Transfer dialplan
application/ARI redirect capability.

*New Feature*
A new operation has been added to the ARI channels resource, redirect. With
this, a channel in a Stasis application can be redirected to another endpoint
of the same underlying channel technology.

*Bug fixes*
In the process of writing this new feature, two bugs were fixed in the PJSIP
stack:
(1) The existing .transfer channel callback had the limitation that it could
    only transfer channels to a SIP URI, i.e., you had to pass
    'PJSIP/sip:foo@my_provider.com' to the dialplan application. While this is
    still supported, it is somewhat unintuitive - particularly in a world full
    of endpoints. As such, we now also support specifying the PJSIP endpoint to
    transfer to.
(2) res_pjsip_multihomed was, unfortunately, trying to 'help' a 302 redirect by
    updating its Contact header. Alas, that resulted in the forwarding
    destination set by the dialplan application/ARI resource/whatever being
    rewritten with very incorrect information. Hence, we now don't bother
    updating an outgoing response if it is a 302. Since this took a looong time
    to find, some additional debug statements have been added to those modules
    that update the Contact headers.

Review: https://reviewboard.asterisk.org/r/4316/

ASTERISK-24015 #close
Reported by: Private Name

ASTERISK-24703 #close
Reported by: Matt Jordan
........

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      29f66b04
  9. Feb 11, 2015
  11. Feb 10, 2015
    • George Joseph's avatar
      res_pjsip_config_wizard: Add ability to auto-create hints. · 49161d8d
      George Joseph authored 
      Looking at the Super Awesome Company sample reminded me that creating hints is 
just plain gruntwork.  So you can now have the pjsip conifg wizard auto-create 
them for you.

Specifying 'hint_exten' in the wizard will create 
'exten => <hint_exten>,hint/PJSIP/<wizard_id>'
in whatever is specified for 'hint_context'.

Specifying 'hint_application' in the wizard will create
'exten => <hint_exten>,1,<hint_application>'
in whatever is specified for 'hint_context'.

The default for 'hint_context' is the endpoint's context.
There's no default for 'hint_application'.  If not specified, no app is added.
There's no default for 'hint_exten'.  If not specified, neither the hint itself 
nor the application will be created.

Some may think this is the slippery slope to users.conf but hints are a basic 
necessity for phones unlike voicemail, manager, etc that users.conf creates.

Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4383/
........

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      49161d8d
  13. Feb 09, 2015
  15. Feb 08, 2015
  17. Feb 06, 2015
  19. Feb 04, 2015
  21. Jan 31, 2015
  23. Jan 30, 2015
  25. Jan 29, 2015
  27. Jan 28, 2015
    • Mark Michelson's avatar
      Fix file descriptor leak in RTP code. · b3ff43a4
      Mark Michelson authored 
      SIP requests that offered codecs incompatible with configured values
could result in the allocation of RTP and RTCP ports that would not get
reclaimed later.

ASTERISK-24666 #close
Reported by Y Ateya

Review: https://reviewboard.asterisk.org/r/4323

AST-2015-001
........

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      b3ff43a4
    • Mark Michelson's avatar
      Multiple revisions 431297-431298 · 3cccfac3
      Mark Michelson authored 
      ........
  r431297 | mmichelson | 2015-01-28 11:05:26 -0600 (Wed, 28 Jan 2015) | 17 lines
  
  Mitigate possible HTTP injection attacks using CURL() function in Asterisk.
  
  CVE-2014-8150 disclosed a vulnerability in libcURL where HTTP request injection
  can be performed given properly-crafted URLs.
  
  Since Asterisk makes use of libcURL, and it is possible that users of Asterisk may
  get cURL URLs from user input or remote sources, we have made a patch to Asterisk
  to prevent such HTTP injection attacks from originating from Asterisk.
  
  ASTERISK-24676 #close
  Reported by Matt Jordan
  
  Review: https://reviewboard.asterisk.org/r/4364
  
  AST-2015-002
........
  r431298 | mmichelson | 2015-01-28 11:12:49 -0600 (Wed, 28 Jan 2015) | 3 lines
  
  Fix compilation error from previous patch.
........

Merged revisions 431297-431298 from http://svn.asterisk.org/svn/asterisk/branches/11
........

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      3cccfac3
    • Sean Bright's avatar
      media formats: update res_format_attr_opus & silk · f080ca65
      Sean Bright authored 
      In r419044, we changed how formats were handled, but the return value
of the format_parse_sdp_fmtp functions in res_format_attr_opus and
res_format_attr_silk were not updated, causing calls to fail.  Ran
into this when getting codec_opus working with Asterisk 13.

Once the return value was corrected, we were crashing in opus_getjoint
because of NULL format attributes.  I've fixed this as well in this
patch.

Review: https://reviewboard.asterisk.org/r/4371/
........

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      f080ca65
    • Richard Mudgett's avatar
      res_pjsip_outbound_registration: Fix reload race condition. · 69e107b2
      Richard Mudgett authored 
      Performing a CLI "module reload" command when there are new pjsip.conf
registration objects defined frequently failed to load them correctly.

What happens is a race condition between res_pjsip pushing its reload into
an asynchronous task processor task and the thread that does the rest of
the reloads when it gets to reloading the res_pjsip_outbound_registration
module.  A similar race condition happens between a reload and the CLI/AMI
show registrations commands.  The reload updates the current_states
container and the CLI/AMI commands call get_registrations() which builds a
new current_states container.

* Made res_pjsip.c reload_module() use ast_sip_push_task_synchronous()
instead of ast_sip_push_task() to eliminate two threads processing config
reloads at the same time.

* Made get_registrations() not replace the global current_states container
so the CLI/AMI show registrations command cannot interfere with reloading.
You could never add/remove objects in the container without the
possibility of the container being replaced out from under you by
get_registrations().

* Added a registration loaded sorcery instance observer to purge any dead
registration objects since get_registrations() cannot do this job anymore.
The struct ast_sorcery_instance_observer callbacks must be used because
the callback happens inline with the load process.  The struct
ast_sorcery_observer callbacks are pushed to a different thread.

* Added some global current_states NULL pointer checks in case the
container disappears because of unload_module().

* Made sorcery's struct ast_sorcery_instance_observer.object_type_loaded
callbacks guaranteed to be called before any struct
ast_sorcery_observer.loaded callbacks will be called.

* Moved the check for non-reloadable objects to before the sorcery
instance loading callbacks happen to short circuit unnecessary work.
Previously with non-reloadable objects, the sorcery instance
loading/loaded callbacks would always happen, the individual wizard
loading/loaded would be prevented, and the non-reloadable type logging
message would be logged for each associated wizard.

ASTERISK-24729 #close
Review: https://reviewboard.asterisk.org/r/4381/
........

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      69e107b2
  29. Jan 27, 2015
    • Kevin Harwell's avatar
      tcptls: Bad file descriptor error when reloading chan_sip · c7591ef6
      Kevin Harwell authored 
      While running through some scenarios using chan_sip and tcp a problem would
occur that resulted in a flood of bad file descriptor messages on the cli:

tcptls.c:712 ast_tcptls_server_root: Accept failed: Bad file descriptor

The message is received because the underlying socket has been closed, so is
valid. This is probably happening because unloading of chan_sip is not atomic.
That however is outside the scope of this patch. This patch simply stops the
logging of multiple occurrences of that message.

ASTERISK-24728 #close
Reported by: Thomas Thompson
Review: https://reviewboard.asterisk.org/r/4380/
........

Merged revisions 431218 from http://svn.asterisk.org/svn/asterisk/branches/11
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      c7591ef6
    • Jonathan Rose's avatar
      Manager: Fix Manager Action ModuleLoad to give correct response when reloading · e826cb8a
      Jonathan Rose authored 
      Prior to this patch, ModuleLoad would respond with an error indicating that
the requested module wasn't found in spite of finding and reloading the
module.

Review: https://reviewboard.asterisk.org/r/4373/
ASTERISK-24721 #close
........

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      e826cb8a
    • Kevin Harwell's avatar
      chan_sip: stale nonce causes failure · 3b0f03ef
      Kevin Harwell authored 
      When refreshing (with a small expiration) a registration that was sent to
chan_sip the nonce would be considered stale and reject the registration.
What was happening was that the initial registration's "dialog" still existed
in the dialogs container and upon refresh the dialog match algorithm would
choose that as the "dialog" instead of the newly created one. This occurred
because the algorithm did not check to see if the from tag matched if
authentication info was available after the 401. So, it ended up assuming
the original "dialog" was a match and stopped the search. The old "dialog"
of course had an old nonce, thus the stale nonce message.

This fix attempts to leave the original functionality alone except in the case
of a REGISTER. If a REGISTER is received if searches for an existing "dialog"
matching only on the callid. If the expires value is low enough it will reuse
dialog that is there, otherwise it will create a new one.

ASTERISK-24715 #close
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/4367/
........

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      3b0f03ef
    • Kevin Harwell's avatar
      res_pjsip: make it unloadable (take 2) · e62bd465
      Kevin Harwell authored 
      Due to the original patch causing memory corruptions it was removed until the
problem could be resolved. This patch is the original patch plus some added
locking around stasis router subcription that was needed to avoid the memory
corruption.

Description of the original problem and patch (still applicable):

The res_pjsip module was previously unloadable. With this patch it can now
be unloaded.

This patch is based off the original patch on the issue (listed below) by Corey
Farrell with a few modifications. Namely, removed a few changes not required to
make the module unloadable and also fixed a bug that would cause asterisk to
crash on unloading.

This patch is the first step (should hopefully be followed by another/others at
some point) in allowing res_pjsip and the modules that depend on it to be
unloadable. At this time, res_pjsip and some of the modules that depend on
res_pjsip cannot be unloaded without causing problems of some sort.

The goal of this patch is to get res_pjsip and only res_pjsip to be able to
unload successfully and/or shutdown without incident (crashes, leaks, etc...).
Other dependent modules may still cause problems on unload.

Basically made sure, with the patch applied, that res_pjsip (with no other
dependent modules loaded) could be succesfully unloaded and Asterisk could
shutdown without any leaks or crashes that pertained directly to res_pjsip.

ASTERISK-24485 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4363/
patches:
  pjsip_unload-broken-r1.patch submitted by Corey Farrell (license 5909)
........

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      e62bd465