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There is a reason the heap comparison functions like qsort(), and other
comparison functions specify <0, >0, and =0 for the return values.


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When you have lots of SIP peers (according to the issue reporter, around 3500),
the 'sip show peers' CLI command or AMI action can crash due to a poorly placed
string duplication that occurs on the stack. This patch refactors the command
to not allocate the string on the stack, and handles the formatting of a single
peer in a separate function call.

(closes issue ASTERISK-21466)
Reported by: Guillaume Knispel
patches:
  fix_sip_show_peers_stack_overflow_asterisk_11.3.0-v2.patch uploaded by gknispel (License 6492)
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Way back when in the dark days of Asterisk 1.8.9, blind transferring a call
in a context that included the 'h' extension would inadvertently execute the
hangup code logic on the transferred channel. This was a "bad thing". The fix
was to properly check for the softhangup flags on the channel and only execute
the 'h' extension logic (and, in later versions, hangup handler logic) if the
channel was well and truly dead (Jim).

Unfortunately, CDRs are fickle. Setting the softhangup flag when we detected
that the channel was leaving the bridge (but not to die) caused some crucial
snippet of CDR code, lying in ambush in the middle of the bridging code, to
not get executed. This had the effect of blowing away one of the CDRs that is
typically created during a blind transfer.

While we live and die by the adage "don't touch CDRs in release branches", this
was our bad. The attached patch restores the CDR behavior, and still manages to
not run the 'h' extension during a blind transfer (at least not when it's
supposed to).

Thanks to Steve Davies for diagnosing this and providing a fix.

Review: https://reviewboard.asterisk.org/r/2476

(closes issue ASTERISK-21394)
Reported by: Ishfaq Malik
Tested by: Ishfaq Malik, mjordan
patches:
  fix_missing_blindXfer_cdr2 uploaded by one47 (License 5012)
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(issue ASTERISK-21103)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2481/


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16 bit signed integers have a range of [-32768, 32768).  The existing code
was using the interval (-32768, 32768) instead.  This patch fixes that.

Review: https://reviewboard.asterisk.org/r/2479/
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* Fixed crash when res_stasis_http is unloaded before the
  implementation modules.
* Cleaned up test initialization for test_stasis_http.so.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386928 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1.4.24 core sounds includes a full set of Italian prompts for core sounds and a fix for the missing voicemail prompts in the Russian language.

(closes issue ASTERISK-19431)
(closes issue ASTERISK-19721)
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Review: https://reviewboard.asterisk.org/r/2263/
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This change does the following:

1. Adds the sorcery realtime module
2. Adds unit tests for the sorcery realtime module
3. Changes the realtime core to use an ast_variable list instead of variadic arguments
4. Changes all realtime drivers to accept an ast_variable list

Review: https://reviewboard.asterisk.org/r/2424/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386731 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch updates the various res_sip modules with their proper menuselect
options and proper dependencies, such that Asterisk still has a snowball's
chance in hell of compiling without pjproject.

Much thanks to snuffy(-home|-work) for making everyone's life
easier with this patch.

Review: https://reviewboard.asterisk.org/r/2472/

(closes issue ASTERISK-21669)
Reported by: snuffy
patches:
  xml-depends.diff uploaded by snuffy (license 5024)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386686 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If a system allows for its usage, Asterisk will use glob to help parse
Asterisk .conf files. The config file loading routine was leaking the memory
allocated by the glob() routine when the config file was in an unmodified
or invalid state.

This patch properly calls globfree in those off nominal paths.

(closes issue ASTERISK-21412)
Reported by: Corey Farrell
patches:
  config_glob_leak.patch uploaded by Corey Farrell (license 5909)
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But now it only prints during the initial startup, and prints at verbose 1
level.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386684 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch cleans up two things features:
* It properly unregisters the CLI commands that features registered
* It cancels and performs a pthread_join on the created parking thread. This
  not only properly joins a non-detached thread, but also prevents disposing
  of the parking lots prior to the parking thread completely exiting.

(closes issue ASTERISK-21407)
Reported by: Corey Farrell
patches:
  features_shutdown-r2.patch uploaded by Corey Farrell (License 5909)
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Thanks to Olle Johansson for suggesting this.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386638 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In order to get people familiar with the Stasis message bus, it would
be useful to have something of a tutorial. Since I'm not clever enough
to think of some cool integration we could do with Twitter, I settled
for something that might actually be useful.

This patch adds a res_statsd.so module, which implements a basic
statsd[1] client. Statsd is a very simple statistics gathering server,
which can publish its results to a backend graphing engine, like
Graphite[2]. There are several different Statsd server
implementations[3], so you can pick what works best for your
environment.

The actual example of how to use the Stasis message bus is in
res_chan_stats.so. This module demonstrates how to use subscriptions
and the message router by monitoring messages and posting channels
stats to the statsd server.

A wiki page walking through res_chan_stats.so is forthcoming.

 [1]: https://github.com/etsy/statsd/
 [2]: http://graphite.readthedocs.org/en/latest/
 [3]: http://joemiller.me/2011/09/21/list-of-statsd-server-implementations/

Review: https://reviewboard.asterisk.org/r/2460/


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Don't bind to anything in the sample configuration so we don't clash with chan_sip on a "make samples" right now.


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The pimp_my_sip branch is being merged at this point because
it offers basic functionality, and from an API standpoint, things
are complete.

SIP work is *not* feature-complete; however, with the completion
of the SUBSCRIBE/NOTIFY API, all APIs (except a PUBLISH API) have
been created, and thus it is possible for developers to attempt
to create new SIP work.

API documentation can be found in the doxygen in the code, but
usability documentation is still lacking.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386540 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Use comedia_string() to display correctly the symmetric rtp setting when
  running "sip show settings"
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* Change "ForcerPort" to "Forcerport" to match everywhere else it is displayed
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* Make confbridge config parsing user profile, bridge profile, and menu
container hash/cmp functions correctly check the OBJ_POINTER, OBJ_KEY, and
OBJ_PARTIAL_KEY flags.

* Made confbridge load_module()/unload_module() free all resources on
failure conditions.


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This crept in with the RESTful HTTP interface merge.


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The two party bridging loops were changing the bridge peer pointers
without the channel locks held.  Thus when ast_channel_massquerade()
tested and used the pointer there is a small window of opportunity for the
pointers to become NULL even though the masquerade code has the channels
locked.

(closes issue ASTERISK-21356)
Reported by: William luke
Patches:
      jira_asterisk_21356_v11.patch (license #5621) patch uploaded by rmudgett
Tested by: William luke
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