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Author SHA1 Message Date
Joshua Colp
47074f4bfd res_pjsip: Ensure in-dialog responses have an endpoint associated. 
When handling incoming messages we determine if it is associated with
a dialog. If so we use that to determine what serializer and endpoint
to use for the message. Previously this would pass the endpoint to the
endpoint lookup module to actually place the endpoint completely on the
message. For in-dialog responses, however, this did not occur as
dialog processing took over and the endpoint lookup did not occur.

This change just places the endpoint in the expected spot immediately
instead of relying on the endpoint lookup module. In-dialog responses
thus have the expected endpoint.

AST-1459 #close

Review: https://reviewboard.asterisk.org/r/4146/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427492 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-11-06 18:21:12 +00:00
Corey Farrell
4d80f223af main/file.c: fix possible extra ast_module_unref to format modules. 
fn_wrapper only adds a reference to the format's module if the file
was able to be opened.  If not this causes an unmatched
ast_module_unref in filestream_destructor.  Move ast_module_ref to
get_stream.

ASTERISK-24492 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4149/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427467 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-11-06 12:15:54 +00:00
Corey Farrell
c46664305a res_hep: fix major leak that occurs when config is missing or enabled=no. 
Add missing unreference in hepv3_send_packet.

ASTERISK-24491 #close
Reported by: Zane Conkle
Tested by: Zane Conkle
Review: https://reviewboard.asterisk.org/r/4150/
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 2014-11-06 09:24:26 +00:00
Corey Farrell
7e2369310c Fix unintential memory retention in stringfields. 
* Fix missing / unreachable calls to __ast_string_field_release_active.
* Reset pool->used to zero when the current pool->active reaches zero.

ASTERISK-24307 #close
Reported by: Etienne Lessard
Tested by: ibercom, Etienne Lessard
Review: https://reviewboard.asterisk.org/r/4114/
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 2014-11-06 09:18:48 +00:00
George Joseph
362dde2229 test_strings: Remove string tests that exercise asserts. 
Since unit tests are run with DO_CRASH, those tests were causing
the test to fail.

Tested-by: George Joseph
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 2014-11-06 02:41:17 +00:00
Mark Michelson
69f29e627f Make the disable_tcp_switch PJSIP system object enabled by default. 
Testing has shown repeatedly that PJSIP's default behavior of switching
automatically to TCP for large messages can cause issues. The most common
issues are that devices that we are communicating with do not handle the
switch to TCP gracefully, thus causing situations such as broken calls or
broken subscriptions. Now, in order to have this behavior happen, you must
opt into it. The sample file has been updated to warn that enabling the
TCP switch behavior may cause issues for you, so use at your own risk.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427335 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-11-05 19:53:29 +00:00
Joshua Colp
b06078880b res_pjsip_multihomed: Add logging during startup to aid debugging if local DNS is misbehaving. 
This change adds a bit of logging so if the local DNS is misbehaving it is easier
to track down what is going on and where Asterisk may be hanging.

ASTERISK-24438 #close
Reported by: Melissa Shepherd

Review: https://reviewboard.asterisk.org/r/4148/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427306 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-11-05 12:19:09 +00:00
George Joseph
d5de94201e config: Make text_file_save and 'dialplan save' escape semicolons in values. 
When a config file is read, an unescaped semicolon signals comments which are
stripped from the value before it's stored.  Escaped semicolons are then
unescaped and become part of the value.  Both of these behaviors are normal
and expected.  When the config is serialized either by 'dialplan save' or
AMI/UpdateConfig however, the now unescaped semicolons are written as-is.
If you actually reload the file just saved, the unescaped semicolons are
now treated as start of comments.

Since true comments are stripped on read, any semicolons in
ast_variable.value must have been escaped originally.  This patch
re-escapes semicolons in ast_variable.values before they're written to
file either by 'dialplan save' or config/ast_config_text_file_save which
is called by AMI/UpdateConfig. I also fixed a few pre-existing formatting
issues nearby in pbx_config.c

Tested-by: George Joseph
ASTERISK-20127 #close

Review: https://reviewboard.asterisk.org/r/4132/
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 2014-11-05 00:17:10 +00:00
Joshua Colp
c77a71ad2f res_pjsip: Apply the 'user_eq_phone' setting to the To header as well. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427259 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-11-04 22:51:32 +00:00
Joshua Colp
5e43d68717 res_pjsip: Allow + at the beginning of a phone number when user_eq_phone is enabled. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427257 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-11-04 22:31:16 +00:00
George Joseph
bd42a09d7f config: BUG: Restore ability for non-templ to be used as base objs in config. 
My recent refactor of config.c accidentally removed the capability for an
object to inherit from a non-template object.

This patch restores the capability to inherit from both template and
non-template objects.

Tested-by: George Joseph
Reported-by: Scott Griepentrog
ASTERISK-24487 #close

Review: https://reviewboard.asterisk.org/r/4147/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427229 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-11-04 20:49:14 +00:00
Corey Farrell
97e1c7f3a9 func_talkdetect: Fix stasis message leak in audiohook callback. 
ASTERISK-24482 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4142/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427205 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-11-04 19:46:33 +00:00
Corey Farrell
9f2874639d res_http_websockets: Fix extra unref of module 
In websocket_add_protocol_internal is used to add the "echo"
protocol, but ast_websocket_remove_protocol is used to remove
it.  This causes an extra call to ast_module_unref.

ASTERISK-24480 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4140/
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 2014-11-04 19:33:21 +00:00
Corey Farrell
bdc35c77b9 Fix crash caused by merge error on review 4138 
When merging from 12 to 13 there were conflicts,
I mistakenly had the loop run ast_closestream(others[0])
when it should be ast_closestream(others[x]).
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-11-04 14:11:54 +00:00
Joshua Colp
d159885e50 res_pjsip_outbound_registration: Add virtual line support. 
Virtual line support establishes a relationship between messages
related to an outbound registration and a local endpoint. This is
accomplished by attaching a parameter to the Contact of the outbound
registration and looking for it on any received requests. If the
parameter exists and can be matched to an outbound registration
the configured endpoint is associated with the request.

Review: https://reviewboard.asterisk.org/r/2964/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427165 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-11-04 12:03:35 +00:00
Richard Mudgett
33f0251b6c res_pjsip: Add disable_tcp_switch option. 
When a packet exceeds the MTU, pjproject will switch from UDP to TCP.  In
some circumstances (on some networks), this can cause some issues with
messages not getting sent to the correct destination - and can also cause
connections to get dropped due to quirks in pjproject deciding to
terminate TCP connections with no messages.

While fixing the routing/messaging issues is important, having a
configuration option in Asterisk that tells pjproject to not switch over
to TCP would be useful.  That way, if some glitch is discovered on some
other network/site, we can at least disable the behavior until a fix is
put into place.

AFS-197 #close

Review: https://reviewboard.asterisk.org/r/4137/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427137 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-11-03 18:22:59 +00:00
Joshua Colp
b9aeff9580 chan_pjsip: Update CHANGES file to include 'moh_passthrough' setting 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427113 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-11-03 15:03:38 +00:00
Joshua Colp
ac091d4184 chan_pjsip: Add support for passing hold and unhold requests through. 
This change adds an option, moh_passthrough, that when enabled will pass
hold and unhold requests through using a SIP re-invite. When placing on
hold a re-invite with sendonly will be sent and when taking off hold a
re-invite with sendrecv will be sent. This allows remote servers to handle
the musiconhold instead of the local Asterisk instance being responsible.

Review: https://reviewboard.asterisk.org/r/4103/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427112 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-11-03 14:45:01 +00:00
Corey Farrell
285be15aaf Fix compile error caused by review 4138 
There is no procedure called ast_closeframe, fix code to use
ast_closestream.

Reported By: Matt Jordan
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 2014-11-03 02:36:46 +00:00
Corey Farrell
509c04ef38 Fix ast_writestream leaks 
Fix cleanup in __ast_play_and_record where others[x] may be leaked.
This was caught where prepend != NULL && outmsg != NULL, once
realfile[x] == NULL any further others[x] would be leaked. A cleanup
block was also added for prepend != NULL && outmsg == NULL.

11+: Fix leak of ast_writestream recording_fs in
app_voicemail:leave_voicemail.

ASTERISK-24476 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4138/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427027 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-11-02 08:13:52 +00:00
Corey Farrell
85c1822a9d func_jitterbuffer: fix frame leaks. 
Fix code paths where it is possible for frames to leak.
Fix uninitialized variable in jb_get_fixed and jb_get_adaptive.

ASTERISK-22409 #related
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4128/
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 2014-11-02 07:40:57 +00:00
Matthew Jordan
5db1c978e3 res/res_stasis: Fix crash on module unload while performing operation 
When the res_stasis module is unloaded, it will dispose of the apps_registry
container. This is a problem if an ARI operation is in flight that attempts
to use the registry, as the shutdown occurs in a separate thread. This patch
adds some sanity checks to the various routines that access the registry which
cause the operations to fail if the apps_registry does not exist.

Crash caught by the Asterisk Test Suite.
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 2014-11-02 01:01:52 +00:00
Tzafrir Cohen
4219c40775 install init.d files on GNU/kFreeBSD 
Review: https://reviewboard.asterisk.org/r/4118/
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 2014-10-31 16:52:42 +00:00
Scott Griepentrog
28173ddf05 pjsip: clarify tls cert and key file usage 
A question arose as to whether a .pem file
could be provided in place of the .crt and
.key files in a PJSIP TLS configuration. I
tested this and discovered that although a
cert will be read from the pem file, a key
will not, and thus the priv_key_file entry
is still required. This update to the fine
documentation clarifies the option usage.

AST-1448 #close
Review: https://reviewboard.asterisk.org/r/4129/
Reported by: John Bigelow
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426932 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-10-31 16:41:06 +00:00
Scott Griepentrog
f59db388a7 pjsip: Handle outbound unregister correctly 
This updates the status of the outbound registration
to reflect when it has been unregistered.  Since the
registration is unregistered but is not stopped, the
registration schedule remains active as before.  The
patch also updates the documentation of both the AMI
and CLI commands.

ASTERISK-24411 #close
Review: https://reviewboard.asterisk.org/r/4119/
Reported by: John Bigelow
patches:
  unregister-patch1.txt uploaded by John Bigelow (License 5091)
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 2014-10-31 16:24:00 +00:00
Matthew Jordan
d88282af40 channels/sip/reqresp_parser: Fix unit tests for r426594 
When r426594 was made, it did not take into account a unit test that verified
that the function properly populated the unsupported buffer. The function
would previously memset the buffer if it detected it had any contents; since
this function can now be called iteratively on successive headers, the unit
tests would now fail. This patch updates the unit tests to reset the buffer
themselves between successive calls, and updates the documentation of the
function to note that this is now required.
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 2014-10-31 03:26:28 +00:00
Corey Farrell
bf684b63a3 REF_DEBUG: Install refcounter.py to $(ASTDATADIR)/scripts 
This change ensures refcounter.py is installed to a place where it
can be found by the Asterisk testsuite if REF_DEBUG is enabled.

ASTERISK-24432 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4094/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426834 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-10-31 03:09:48 +00:00
Corey Farrell
e4374a3abe app_queue: fix a couple leaks to struct call_queue in set_member_value 
set_member_value has a couple leaks to references in the variable q
found through testsuite tests/queues/set_penalty.  Also remove the
REF_DEBUG_ONLY_QUEUES compiler declaration, this is no longer possible
with the updated REF_DEBUG code.

ASTERISK-24466 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4125/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426808 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-10-30 23:56:39 +00:00
Corey Farrell
ced81afff2 audiohooks: Clean references to formats 
Cleanup references to in_translate[x].format and
out_translate[x].format in ast_audiohook_detach_list.

ASTERISK-24465 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4124/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426804 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-10-30 23:45:25 +00:00
Kevin Harwell
a537e314d1 res_pjsip_exten_state: PJSIPShowSubscriptionsInbound causes crash 
Currently, it is possible for some subscriptions to get into a NULL state. When
this occurs and the PJSIPShowSubscriptionsInbound ami action is issued and a
device is subscribed for extension state then the associated subscription state
object can't be located.  The code then attempts to dereference a NULL object.
Added a NULL check to avoid the problem.

Reported by: John Bigelow
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426781 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-10-30 21:14:01 +00:00
Kevin Harwell
cd52456ea1 res_pjsip: incorrect qualify statistics after disabling for contact 
When removing the qualify_frequency from an AoR or a contact the statistics
shown when issuing "pjsip show aors" from the CLI are incorrect. This patch
deletes the contact's status object from sorcery, disassociating it from the
contact, if the qualify_freqency is removed from configuration.

ASTERISK-24462 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4116/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426761 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-10-30 17:18:47 +00:00
Walter Doekes
5d8d90c402 app_voicemail: Fix unchecked bounds of myArray in IMAP_STORAGE. 
In update_messages_by_imapuser(), messages were appended to a finite
array which resulted in a crash when an IMAP mailbox contained more
than 256 entries. This memory is now dynamically increased as needed.

Observe that this patch adds a bunch of XXX's to questionable code. See
the review (url below) for more information.

ASTERISK-24190 #close
Reported by: Nick Adams
Tested by: Nick Adams

Review: https://reviewboard.asterisk.org/r/4126/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426706 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-10-30 09:21:42 +00:00
Igor Goncharovskiy
c866ced76b Add additional checks for NULL pointers to fix several crashes reported. 
ASTERISK-24304 #close
Reported by: dhanapathy sathya
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426669 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-10-30 06:15:14 +00:00
Matthew Jordan
0ddc3bde24 channels/chan_sip: Add improved support for 4xx error codes 
This patch adds support for 414, 493, 479, and a stray 400 response in REGISTER
response handling. This helps interoperability in a number of scenarios.

Review: https://reviewboard.asterisk.org/r/3437

patches:
  rb3437.patch uploaded by oej (License 5267)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426603 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-10-30 01:59:39 +00:00
Matthew Jordan
ff83ff564c channels/chan_sip: Support mutltiple Supported and Required headers 
A SIP request may contain multiple Supported: and Required: headers. Currently,
chan_sip only parses the first Supported/Required header it finds. This patch
adds support for multiple Supported/Required headers for INVITE requests.

Review: https://reviewboard.asterisk.org/r/2478

ASTERISK-21721 #close
Reported by: Olle Johansson
patches:
  rb2478.patch uploaded by oej (License 5267)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426598 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-10-30 01:48:00 +00:00
Tzafrir Cohen
8a69aedd17 Fix building chan_phone on big endian systems 
A left over from the formats conversion (Corey Farrell).

ASTERISK-24458 #close
Review: https://reviewboard.asterisk.org/r/4117/

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426573 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-10-29 13:02:27 +00:00
Richard Mudgett
0ed8aebda9 bridge_builtin_features: Add missing channel locks around ast_get_chan_features_general_config(). 
The feature_automonitor() and feature_automixmonitor() functions were not
locking the channel around ast_get_chan_features_general_config().
Accessing the channel datastore list without the channel locked is a good
way to corrupt the list or follow the pointer chain into oblivion.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426553 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-10-28 21:35:41 +00:00
Corey Farrell
7205d76d7d res_fax: Resolve T38 gateway frame leak. 
When frames are translated by a fax gateway they need to be freed.  The
existing call to ast_frfree was unreachable.  This change reorganizes
fax_gateway_framehook to ensure that ast_frfree is called when needed.

ASTERISK-24457 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4115/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426530 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-10-28 21:10:42 +00:00
Corey Farrell
67e496c275 manager: Unsubscribe from acl_change_sub at shutdown. 
ASTERISK-24453 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4110/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426526 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-10-28 20:44:22 +00:00
Malcolm Davenport
1fe22c411d ASTERISK-23512, correct inaccurate comment in manager.conf.sample 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426462 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-10-28 18:09:32 +00:00
Matthew Jordan
8e9f593e3a main/bridge: Destroy features struct on off nominal path during bridge impart 
When a channel is imparted to a bridge, the invocation of the function may
provide an ast_bridge_features struct. Upon passing this to ast_bridge_impart,
the caller must assume that ownership has passed to the function, as in all
paths the function destroys the struct prior to returning (as its purpose is
to configure the behavior of the channel while in the bridge). On one off
nominal path - where the channel already has a PBX thread - the struct was not
being destroyed.

This patch fixes that glitch.

ASTERISK-24437 #close
Reported by: Scott Griepentrog
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426433 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-10-28 16:41:17 +00:00
Matthew Jordan
f4b4d42630 main/manager: Fix typo in AMI event documentation of "OriginateResponse" 
The parameter name is "Response", not "Resonse".

ASTERISK-24430 #close
Reported by: Dafi Ni
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 2014-10-28 14:59:47 +00:00
Malcolm Davenport
68d9872f58 ASTERISK-24323, fix bug in documentation of AGI STREAM FILE CONTROL 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-10-28 14:57:01 +00:00
Malcolm Davenport
684b8762a9 ASTERISK-24419, fix incorrect syntax for setting language in extensions.conf.sample 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426297 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-10-28 13:13:16 +00:00
Corey Farrell
2290393273 app_queue: Cleanup ao2_iterator 
Clean ao2_iterator, resolving reference leak to queue members.

ASTERISK-24454 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4111/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-10-28 11:22:55 +00:00
Corey Farrell
ab16f46139 func_cdr: Fix CDR_PROP payload leak 
Remove duplicate allocation of payload, preventing leak.

ASTERISK-24455 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4113/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426253 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-10-28 11:12:03 +00:00
Sean Bright
ef8cdd40e5 configure: Add autoconf check for libopus. 
Because opus transcoding support cannot be included in the standard Asterisk
distribution, a few codec_opus implementations have popped up.  To make it
easier for people to drop in opus support in their own installations, this
patch adds configure checks for libopus.

Review: https://reviewboard.asterisk.org/r/4106/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426235 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-10-27 17:55:43 +00:00
Matthew Jordan
5a17878085 res/res_http_websocket: Fix minor nits found by wdoekes on r409681 
When Moises committed the fixes for WSS (which was a great patch), wdoekes had
a few style nits that were on the review that got missed. This patch resolves
what I *think* were all of the ones that were still on the review.

Thanks to both moy for the patch, and wdoekes for the reviews.

Review: https://reviewboard.asterisk.org/r/3248/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-10-27 02:47:03 +00:00
Matthew Jordan
62bee9b327 res/res_phoneprov: Fix crash on shutdown caused by container cleanup 
In res_phoneprov, unloading the module first destroys the http_routes
container, followed by the users. However, users may have a route in
the http_routes container; the validity of this container is not checked
in the users destructor. Hence, we hit an assert as the container has already
been set to NULL.

This patch does two things:
(1) It adds a sanity check in the user destructor (because why not)
(2) It switches the order of destruction, so that users are disposed of prior
    to the HTTP routes they may hold a reference to.

Note that this crash was caught by the Test Suite (go go testing!)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426179 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2014-10-27 02:27:56 +00:00
Matthew Jordan
130a3fcd7f res/res_srtp: Fix include issue for libsrtp 1.5.0 
In libsrtp 1.5.0, crypto_get_random is no longer resolved simply by including
srtp.h. Now, one must include crypto_kernel.h as well. As it turns out, this
header file has been provided by the library since 2006, so this is a
relatively benign change.

ASTERISK-24436 #close
Reported by: Patrick Laimbock
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 2014-10-27 01:47:56 +00:00