asterisk/asterisk.git
7 years agores_pjsip: crash when using localnet and external_signaling_address options
Kevin Harwell [Fri, 27 Sep 2013 18:28:41 +0000 (18:28 +0000)]
res_pjsip: crash when using localnet and external_signaling_address options

There was a collision of mod_data use on the transaction between using a nat
hook and an session response callback.  During state change it was assumed
what was in the mod_data was nothing or the response callback.  However, it
was possible for it to also contain a nat hook thus resulting in a bad cast
and a crash.

Added the ability to store multiple data elements in mod_data via a hash table.
In this instance, mod_data now stores a hash table of the two values that can
be retrieved using an associated string key.

(closes issue ASTERISK-22394)
Reported by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/2843/
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7 years agochan_sip: Reject calls on 200 OKs if no SDP has been received
Jonathan Rose [Fri, 27 Sep 2013 17:46:16 +0000 (17:46 +0000)]
chan_sip: Reject calls on 200 OKs if no SDP has been received

When Asterisk receives a 200 OK in response to an invite, that peer should have
sent an SDP at some point by then. If the channel has never received an SDP,
media won't have been set and the remote address won't be known. Endpoints in
general should not be doing this. This patch makes it so that Asterisk will
simply hang up a call if it sends a 200 OK at this point. So far this odd
behavior for endpoints has only been observed in tests which involved manually
created SIP transactions in SIPp.

(closes issue ASTERISK-22424)
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/2827/
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7 years agoastobj2: Remove OBJ_CONTINUE support.
Richard Mudgett [Fri, 27 Sep 2013 17:11:22 +0000 (17:11 +0000)]
astobj2: Remove OBJ_CONTINUE support.

OBJ_CONTINUE was a strange feature that came into the world under
suspicious circumstances to support an abuse of the ao2_container by
chan_iax2.  Since chan_iax2 no longer uses OBJ_CONTINUE, it is safe to
remove it.

The simplified code should help performance slightly and make
understanding the code easier.

Review: https://reviewboard.asterisk.org/r/2887/
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7 years agoFix refleaks of ast_rtp_instance structures.
Mark Michelson [Fri, 27 Sep 2013 14:35:12 +0000 (14:35 +0000)]
Fix refleaks of ast_rtp_instance structures.

These refleaks were causing bridged calls not to close their RTP ports. Thus
a call would leave open 4 ports (RTP for party A, RTCP for party A, RTP for party
B, and RTCP for party B). This led to an eventual depletion of available RTP
ports.
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7 years agoRestore usefulness of the CEL Peer field
Kinsey Moore [Fri, 27 Sep 2013 14:08:23 +0000 (14:08 +0000)]
Restore usefulness of the CEL Peer field

This change makes the CEL peer field useful again for BRIDGE_ENTER and
BRIDGE_EXIT events and fills the field with a comma-separated list of
all channels in the bridge other than the channel that is entering or
exiting the bridge.

Review: https://reviewboard.asterisk.org/r/2840/
(closes issue ASTERISK-22393)
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7 years agopjsip: race condition in registrar
Kevin Harwell [Thu, 26 Sep 2013 18:51:54 +0000 (18:51 +0000)]
pjsip: race condition in registrar

While handling a registration request a race condition could occur if/when two+
clients registered at the same time.  This happened when one request obtained a
copy of the current contacts for an AOR and another request did the same before
the first request updated.  Thus the second would update and overwrite the first
(or vice-versa depending on which actually updated first).  In the case of it
being the same contact two "add" events would be raised.

pjsip registration handling is now serialized to alleviate this issue.

(closes issue AST-1213)
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/2860/
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7 years agoAdding a few words to the Dial option 'r' help text to clarify its tone argument...
Rusty Newton [Thu, 26 Sep 2013 14:13:37 +0000 (14:13 +0000)]
Adding a few words to the Dial option 'r' help text to clarify its tone argument description
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7 years agochan_dahdi: CLI "core stop gracefully" has needless delay for PRI and SS7.
Richard Mudgett [Wed, 25 Sep 2013 20:38:24 +0000 (20:38 +0000)]
chan_dahdi: CLI "core stop gracefully" has needless delay for PRI and SS7.

The PRI and SS7 link control threads are not stopped correctly when the
chan_dahdi.so module is unloaded.  The link control threads pri_dchannel()
and ss7_linkset() are not awakened from a poll() to cancel the thread.

* Added a SIGURG signal after requesting the thread cancel to break the
link control thread poll() immediately.

For SS7 it was slightly worse, the link poll() timeout would always be
whatever was the last libss7 scheduled event time used.  If no libss7
scheduled event was pending, the thread could run more often than
necessary.

* Set nextms to 60 seconds for the ss7_linkset() poll() if there is no
other libss7 scheduled event.
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7 years agoBroke the build - Fixing XML DTD violation added in r399782, missing <para> tags...
Rusty Newton [Wed, 25 Sep 2013 19:43:43 +0000 (19:43 +0000)]
Broke the build - Fixing XML DTD violation added in r399782, missing <para> tags inside a <note>
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7 years agochan_sip: Fix Realtime Peer Update Problem When Un-registering And Expires Header...
Michael L. Young [Wed, 25 Sep 2013 19:29:38 +0000 (19:29 +0000)]
chan_sip: Fix Realtime Peer Update Problem When Un-registering And Expires Header In 200ok

1st Issue
When a realtime peer sends an un-REGISTER request, Asterisk
un-registers the peer but the database table record still has regseconds and
fullcontact for the peer.  This results in calls attempting to be routed to the
peer which is no longer registered.  The expected behavior is to get
busy/congested when attempting to call an un-registered peer through the
dialplan.

What was discovered is that we are clearing out the peer's registration in the
database in parse_register_contact() when calling expire_register() but then
upon returning from parse_register_contact(), update_peer() is run which stores
back in the database table regseconds and fullcontact.

2nd Issue
The reporter pointed out that the 200 ok being returned by Asterisk
after un-registering a peer contains a Contact header with ;expires= and the
Expires header is not set to 0.  This is actually a regression.

Tests were created for this second issue (ASTERISK-22548).  The tests have been
reviewed and a Ship It! was received on those tests.

This patch does the following:

* Do not ignore the Expires header value even when it is set to 0.  The patch
  sets the pvt->expiry earlier on in the function so that it is set properly and
  used.

* If pvt->expiry is 0, do not call update_peer since that means the peer has
  already been un-registered and there is no need to update the database record
  again since nothing has changed.

(closes issue ASTERISK-22428)
Reported by: Ben Smithurst
Tested by: Ben Smithurst, Michael L. Young
Patches:
  asterisk-22428-rt-peer-update-and-expires-header.diff
                                              by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2869/
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7 years agoFixing documentation for the configOption "external_media_address" of both Endpoints...
Rusty Newton [Wed, 25 Sep 2013 18:38:34 +0000 (18:38 +0000)]
Fixing documentation for the configOption "external_media_address" of both Endpoints and Transports

Re-using some of Mark Michelson's text from an E-mail discussion for:

* Modifying synopsis for both options
* Adding description to both options
* Changing name of "external_media_address" for Endpoint configuration to "media_address" in anticipation of the option name being changed. (As it is not really specific to external destinations)

(issue ASTERISK-22405)
(closes issue ASTERISK-22405)
Reported by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/2850/
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7 years agoastobj2: Made use OBJ_SEARCH_xxx identifiers as field enum values internally.
Richard Mudgett [Tue, 24 Sep 2013 22:55:06 +0000 (22:55 +0000)]
astobj2: Made use OBJ_SEARCH_xxx identifiers as field enum values internally.

* Made ao2_unlink to protect itself from stray OBJ_SEARCH_xxx values
passed in.
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7 years agochan_iax2: Prevent some needless breaking of the native IAX2 bridge.
Richard Mudgett [Tue, 24 Sep 2013 20:37:32 +0000 (20:37 +0000)]
chan_iax2: Prevent some needless breaking of the native IAX2 bridge.

* Clean up some twisted code in the iax2_bridge() loop.

* Add AST_CONTROL_VIDUPDATE and AST_CONTROL_SRCCHANGE to a list of frames
to prevent the native bridge loop from breaking.

* Passing the AST_CONTROL_T38_PARAMETERS frame should also allow FAX over
a native IAX2 bridge.

(issue ABE-2912)

Review: https://reviewboard.asterisk.org/r/2870/
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For v12 and above this is really just documentation until IAX2 native
bridging is restored.
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7 years agoapp_queue: Don't be quite so aggressive in initializing the array
Matthew Jordan [Tue, 24 Sep 2013 19:22:30 +0000 (19:22 +0000)]
app_queue: Don't be quite so aggressive in initializing the array

We only need the first character.
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7 years agoapp_queue: Initialize array holding MixMonitor exec options
Matthew Jordan [Tue, 24 Sep 2013 18:59:05 +0000 (18:59 +0000)]
app_queue: Initialize array holding MixMonitor exec options

If the channel variable MONITOR_EXEC is set, app_queue will pass the specified
execution parameters to the MixMonitor application when a queue is recorded.
If that channel variable is not set, the buffer that holds the escaped value
was not being initialized to NULL, and so would be passed to the MixMonitor
application with garbage. Hilarity ensued as app_mixmonitor attempted to
execute gobeldy-gook.
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7 years agoFix a performance problem CDRs
Matthew Jordan [Tue, 24 Sep 2013 18:10:20 +0000 (18:10 +0000)]
Fix a performance problem CDRs

There is a large performance price currently in the CDR engine. We currently
perform two ao2_callback calls on a container that has an entry for every
channel in the system. This is done to create matching pairs between channels
in a bridge.

As such, the portion of the CDR logic that this patch deals with is how we
make pairings when a channel enters a mixing bridge. In general, when a
channel enters such a bridge, we need to do two things:
 (1) Figure out if anyone in the bridge can be this channel's Party B.
 (2) Make pairings with every other channel in the bridge that is not already
     our Party B.

This is a two step process. In the first step, we look through everyone in the
bridge and see if they can be our Party B (single_state_process_bridge_enter).
If they can - yay! We mark our CDR as having gotten a Party B. If not, we keep
searching. If we don't find one, we wait until someone joins who can be our
Party B.

Step 2 is where we changed the logic
(handle_bridge_pairings and bridge_candidate_process). Previously, we would
first find candidates - those channels in the bridge with us - from the
active_cdrs_by_channel container. Because a channel could be a candidate if it
was Party B to an item in the container, the code implemented multiple
ao2_container callbacks to get all the candidates. We also had to store them
in another container with some other meta information. This was rather complex
and costly, particularly if you have 300 Local channels (600 channels!) going
at once.

Luckily, none of it is needed: when a channel enters a bridge (which is when
we're figuring all this stuff out), the bridge snapshot tells us the unique
IDs of everyone already in the bridge. All we need to do is:
 For all channels in the bridge:
   If the channel is us or our Party B that we got in step 1, skip it
   Compare us and the candidate to figure out who is Party A (based on some
       specific rules)
   If we are Party A:
      Make a new CDR for us, append it to our chain, and set the candidate as
          Party B
   If they are Party A:
      If they don't have a Party B:
        Make a new CDR for them, append us to their chain, and us as Party B
      Otherwise:
        Copy us over as Party B on their existing CDR.

This patch does that.

Because we now use channel unique IDs to find the candidates during bridging,
active_cdrs_by_channel now looks up things using uniqueid instead of channel
name. This makes the more complex code simpler; it does, however, have the
drawback that dialplan applications and functions will be slightly slower as
they have to iterate through the container looking for the CDR by name.
That's a small price to pay however as the bridging code will be called a lot
more often.

This patch also does two other minor changes:
 (1) It reduces the container size of the channels in a bridge snapshot to 1.
     In order to be predictable for multi-party bridges, the order of the
     channels in the container must be stable; that is, it must always devolve
     to a linked list.
 (2) CDRs and the multi-party test was updated to show the relationship between
     two dialed channels. You still want to know if they talked - previously,
     dialed channels were always ignored, which is wrong when they have
     managed to get a Party B.

(closes issue ASTERISK-22488)
Reported by: Richard Mudgett

Review: https://reviewboard.asterisk.org/r/2861/
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7 years agoFix crash in res_pjsip on load if error occurs, and prevent unloading of res_pjsip...
Joshua Colp [Mon, 23 Sep 2013 12:03:18 +0000 (12:03 +0000)]
Fix crash in res_pjsip on load if error occurs, and prevent unloading of res_pjsip and res_pjsip_session.

During load time in res_pjsip if an error occurred the operation would attempt to rollback all
operations done during load. This is not permitted by PJSIP as it will assert if the operation has
not been done. This fix changes the code so it will only rollback what has been initialized already.

Further changes also prevent res_pjsip and res_pjsip_session from being unloaded. This is due to
limitations within PJSIP itself. The library environment can only be changed to a certain extent
and does not provide the ability, currently, to deinitialize certain required functionality.

(closes issue ASTERISK-22474)
Reported by: Corey Farrell
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7 years agores_rtp_asterisk: Fix ref leaks in ast_rtcp_read().
Richard Mudgett [Sat, 21 Sep 2013 04:49:26 +0000 (04:49 +0000)]
res_rtp_asterisk: Fix ref leaks in ast_rtcp_read().

Moved rtcp_report RAII_VAR declaration into the loop so it is unref'ed
after every loop.  Moved message_blob to loop and switched it to a regular
variable.  The regular variable was used since message_blob is used in a
very contained way.

(closes issue ASTERISK-22565)
Reported by: Corey Farrell
Patches:
      rtcp_report-leak.patch (license #5909) patch uploaded by Corey Farrell
Tested by: Corey Farrell
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7 years agomedia_index: Fix process_description_file() memory leak of file_id_persist.
Richard Mudgett [Sat, 21 Sep 2013 01:46:56 +0000 (01:46 +0000)]
media_index: Fix process_description_file() memory leak of file_id_persist.
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7 years agofeatures_config: Fix config ref leak of parkinglots.
Richard Mudgett [Sat, 21 Sep 2013 00:56:52 +0000 (00:56 +0000)]
features_config: Fix config ref leak of parkinglots.

This leak happend for just about every channel created.
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7 years agoapp_queue: Fix json blob ref leak.
Richard Mudgett [Sat, 21 Sep 2013 00:23:45 +0000 (00:23 +0000)]
app_queue: Fix json blob ref leak.

The json ref from queue_member_blob_create() was never released.
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7 years agojson: Make it obvious that ast_json_unref() is NULL safe.
Richard Mudgett [Sat, 21 Sep 2013 00:17:56 +0000 (00:17 +0000)]
json: Make it obvious that ast_json_unref() is NULL safe.

It looked like the safety check was done after the NULL pointer was used.
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7 years agoEnsure global types in the config framework are initialized
Kinsey Moore [Fri, 20 Sep 2013 22:44:11 +0000 (22:44 +0000)]
Ensure global types in the config framework are initialized

If a config object was allocated but one of its global objects was
never encountered, then the global object's defaults were never
applied. Ensure that global objects are initialized properly upon
allocation instead of on configuration.

Review: https://reviewboard.asterisk.org/r/2866/
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7 years agooriginate/call forwarding: Fix a crash when forwarding a call from originate
Jonathan Rose [Fri, 20 Sep 2013 22:06:07 +0000 (22:06 +0000)]
originate/call forwarding: Fix a crash when forwarding a call from originate

(closes issue ASTERISK-22487)
Reported by: David M. Lee
Review: https://reviewboard.asterisk.org/r/2868/
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7 years agoAdd a missing session supplement unregistration in chan_pjsip for ACKs.
Joshua Colp [Fri, 20 Sep 2013 16:18:42 +0000 (16:18 +0000)]
Add a missing session supplement unregistration in chan_pjsip for ACKs.

(closes issue ASTERISK-22453)
Reported by: Corey Farrell
Patches:
chan_pjsip_session_unregister_supplement.patch uploaded by Corey Farrell (license 5909)
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7 years agoFix memory leak in logger.
Kevin Harwell [Fri, 20 Sep 2013 14:26:44 +0000 (14:26 +0000)]
Fix memory leak in logger.

Fixed a memory leak discovered in the logger where a temporary string buffer
was not being freed.

(closes issue ASTERISK-22540)
Reported by: John Hardin
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7 years agooptional_api: Make always use the standard malloc functions even with MALLOC_DEBUG.
Richard Mudgett [Thu, 19 Sep 2013 23:20:43 +0000 (23:20 +0000)]
optional_api: Make always use the standard malloc functions even with MALLOC_DEBUG.
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7 years agochan_sip: Make direct media reinvites for T38 put Asterisk in the media path
Jonathan Rose [Thu, 19 Sep 2013 17:01:09 +0000 (17:01 +0000)]
chan_sip: Make direct media reinvites for T38 put Asterisk in the media path

Prior to this patch, Asterisk would incorrectly use the previous endpoint
addresses in SDP in spite of providing its own port. T38 is never meant to
be done through directmedia and Asterisk should always be in the media path
for these streams.

(closes issue ASTERISK-17273)
Reported by: Kevin Stewart

(closes issue ASTERISK-18706)
Reported by: Jeremy Kister

Review: https://reviewboard.asterisk.org/r/2853/
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7 years agoFix jitter buffer log file creation
Kinsey Moore [Wed, 18 Sep 2013 20:04:14 +0000 (20:04 +0000)]
Fix jitter buffer log file creation

This adjusts '/'-to-'#' replacement to replace all instances of '/'
instead of just the first to ensure that the jitter buffer log file
gets the correct name as per Richard Kenner's suggestion.

(closes issue ASTERISK-21036)
Reported by: Richard Kenner
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7 years agoUpdate prep_tarball with new documentation files on the Asterisk wiki
Matthew Jordan [Wed, 18 Sep 2013 17:23:49 +0000 (17:23 +0000)]
Update prep_tarball with new documentation files on the Asterisk wiki

This will now pull both a command reference for the version being prepared,
as well as an Admin Guide that applies to all versions of Asterisk.

(issue ASTERISK-22439)
Reported by: Olle Johansson
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7 years agoAdd a WARNING in bridge_softmix when a timing module isn't loaded
Matthew Jordan [Wed, 18 Sep 2013 17:21:39 +0000 (17:21 +0000)]
Add a WARNING in bridge_softmix when a timing module isn't loaded

If bridge_softmix fails to be created because no timing source is present in
Asterisk, this will currently fail gracefully but with (most likely) a generic
error message by whatever module tried to create the softmix bridge. This
patch adds a more explicit warning so you can actually diagnose and fix the
problem.

Review: https://reviewboard.asterisk.org/r/2857/
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7 years agoMake config framework able to reload module configs with multiple config files.
Richard Mudgett [Wed, 18 Sep 2013 17:15:53 +0000 (17:15 +0000)]
Make config framework able to reload module configs with multiple config files.

The config framework is supposed to be able to load configs that come from
multiple config files.  The principle example is chan_sip's sip.conf and
users.conf.  Unfortunately, it only does this correctly on initial load.
This patch causes the module's config to be reloaded entirely if any of
the config files change.

(closes issue ASTERISK-22009)
Reported by: Richard Mudgett

Review: https://reviewboard.asterisk.org/r/2859/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399352 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agores_pjsip_messaging: Register message technology as pjsip
Kevin Harwell [Wed, 18 Sep 2013 14:56:22 +0000 (14:56 +0000)]
res_pjsip_messaging: Register message technology as pjsip

pjsip's message technology was being registered as 'sip', which was causing it
to not load due it conflicting with chan_sip's registered 'sip' technology for
messaging.  It now registers as 'pjsip'.  However, due to this change the "to"
field for outgoing pjsip messages need to be prefixed with 'pjsip:' instead of
'sip:'.  Incoming messages to res_pjsip_messaging will automatically have their
"to" fields altered in order to accommodate the change.  Outgoing messages also
handle changing it back to 'sip' before being sent so the pjsip library will
properly handle it.

(closes issue ASTERISK-22445)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2833/
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7 years agoFix Segfault In features-config.c When Application Has No Arguments
Michael L. Young [Wed, 18 Sep 2013 00:13:23 +0000 (00:13 +0000)]
Fix Segfault In features-config.c When Application Has No Arguments

Some applications do not require arguments.  Therefore, when parsing application
maps in features.conf, it is possible that app_data will be set to NULL.

* This patch sets app_data to "" if it is NULL.

Review: https://reviewboard.asterisk.org/r/2804
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7 years agoChange the "external_media_address" PJSIP endpoint option to "media_address".
Mark Michelson [Tue, 17 Sep 2013 23:10:49 +0000 (23:10 +0000)]
Change the "external_media_address" PJSIP endpoint option to "media_address".

The endpoint option does not apply to communication with external entities. Rather,
the option is applied to all communications with the endpoint. The external_media_address
transport configuration option may override the endpoint option if it turns out that
we are going to be communicating with an external entity.

Two things of note:
1) I have not updated the XML documentation. This is being taken care of by Rusty as part
of his work on issue ASTERISK-22405
2) This commit is likely to cause testsuite failures since there are tests that use the
external_media_address endpoint option, and they will need to be changed over. Well, I'm
planning to get that updated ASAP after this commit.

(closes issue ASTERISK-22528)
reported by Rusty Newton
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7 years agoRemote console: more output discrepancies
Kevin Harwell [Tue, 17 Sep 2013 18:44:11 +0000 (18:44 +0000)]
Remote console: more output discrepancies

The remote console continued to have issues with its output.  In this case CLI
command output would either not show up (if verbose level = 0) or would contain
verbose prefixes (if verbose level > 0) once log messages were sent to the
remote console.  The fix now now adds verbose prefix data to all new lines
contained in a verbose log string.

(closes issue ASTERISK-22450)
Reported by: David Brillert
(closes issue AST-1193)
Reported by: Guenther Kelleter
Review: https://reviewboard.asterisk.org/r/2825/
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7 years agoFix doxygen to use correct units of features.conf options.
Richard Mudgett [Tue, 17 Sep 2013 17:55:21 +0000 (17:55 +0000)]
Fix doxygen to use correct units of features.conf options.
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7 years agoFix other timeouts (atxferloopdelay and atxfernoanswertimeout) to use seconds instead...
Mark Michelson [Tue, 17 Sep 2013 17:10:51 +0000 (17:10 +0000)]
Fix other timeouts (atxferloopdelay and atxfernoanswertimeout) to use seconds instead of milliseconds.

Thanks to Richard Mudgett for pointing this out.
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7 years agoSwitch transferdigittimeout to be configured as seconds instead of milliseconds.
Mark Michelson [Tue, 17 Sep 2013 16:11:20 +0000 (16:11 +0000)]
Switch transferdigittimeout to be configured as seconds instead of milliseconds.

This was an unintentional consequence of the update of features.conf to use the
config framework in Asterisk 12. Thanks to Marco Signorini on the Asterisk
developers list for pointing out the problem.
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7 years agoConfbridge: empty conference not being torn down
Kevin Harwell [Tue, 17 Sep 2013 14:58:22 +0000 (14:58 +0000)]
Confbridge: empty conference not being torn down

Confbridge would not properly tear down an empty conference bridge when all
users were kicked via end_marked=yes and at least one user was also set to
wait_marked.  This occurred because while end_marked users were being kicked
and at least one was also set to wait_marked then the leave wait_marked handler
would be called on that user, but there would be no waiting user (still
considered active).  The waiting users would decrement and now be negative.  The
conference would remain, but be put into an inactive state.  The solution was
to move from the active list to the wait list, those users with wait_marked set
right before kicking.  This allows both the active and wait users to decrement
correctly and the confbridge to tear down properly.

A crashed also occurred when trying to list the specific conference from the CLI.
This happened because the conference specified was invalid.  Since the
conference properly tears down now there is no way to reference it thus
alleviating the crash as well.

(closes issue ASTERISK-21859)
Reported by: Chris Gentle
Review: https://reviewboard.asterisk.org/r/2848/
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7 years agoFix module load errors for test_ari_model.so.
Richard Mudgett [Mon, 16 Sep 2013 18:36:22 +0000 (18:36 +0000)]
Fix module load errors for test_ari_model.so.

You cannot use a function pointer variable with an external function from
another dynamically loaded module because data variables are always
resolved even with RTLD_LAZY.

* Added wrapper functions for ast_ari_validate_int() and
ast_ari_validate_string() to use instead for the function pointer
variable.

(closes issue ASTERISK-22457)
Reported by: David M. Lee
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7 years agoapp_speech_utils: Fix unresolved symbol ast_speech_get_setting().
Richard Mudgett [Mon, 16 Sep 2013 18:00:32 +0000 (18:00 +0000)]
app_speech_utils: Fix unresolved symbol ast_speech_get_setting().

Fixes regression introduced by -r374096.

* Made res_speech.export.in export ast_* symbols instead of specific
functions.

* Made app_speech_utils.c declare that it is dependent upon res_speech.

(issue ASTERISK-17136)
Reported by: Richard Kenner
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7 years agochan_iax2: Fix saving the wrong expiry time in astdb.
Richard Mudgett [Mon, 16 Sep 2013 16:50:02 +0000 (16:50 +0000)]
chan_iax2: Fix saving the wrong expiry time in astdb.

When a new IAX2 client registers, the astdb database is updated with the
value of minregexpire defined in iax.conf instead of using the expiry time
that is provided by the client.  The provided expiry time of the client is
updated after inserting the astdb entry.  As a consequence, restarting or
reloading asterisk creates clients whose registration may expire before
they reregister.  The clients are therefore unavailable after minregexpire
seconds until they reregister.

* Move updating of the expiry time to before inserting into the astdb.

(closes issue ASTERISK-22504)
Reported by: Stefan Wachtler
Patches:
      chan_iax2.c.patch (license #6533) patch uploaded by Stefan Wachtler
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7 years agoFilter internal channels out of bridge enter/leave message handling
Matthew Jordan [Mon, 16 Sep 2013 02:37:56 +0000 (02:37 +0000)]
Filter internal channels out of bridge enter/leave message handling

Some channels exist merely as an implementation detail in Asterisk, such as
ConfBridge's announcer/recorder channels. These channels should never be
exposed to the outside world, or to interfaces that report on Asterisk. We
already filter out such channels in snapshot processing; however, we failed to
filter out bridge related messages that involved these channels.

This patch filters out bridge related messages that are for such channels. This
prevents a spurious WARNING message from being displayed when those channels
move in and out of bridges.
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7 years agoRestore Dial, Queue, and FollowMe 'I' option support.
Richard Mudgett [Fri, 13 Sep 2013 22:19:23 +0000 (22:19 +0000)]
Restore Dial, Queue, and FollowMe 'I' option support.

The Dial, Queue, and FollowMe applications need to inhibit the bridging
initial connected line exchange in order to support the 'I' option.

* Replaced the pass_reference flag on ast_bridge_join() with a flags
parameter to pass other flags defined by enum ast_bridge_join_flags.

* Replaced the independent flag on ast_bridge_impart() with a flags
parameter to pass other flags defined by enum ast_bridge_impart_flags.

* Since the Dial, Queue, and FollowMe applications are now the only
callers of ast_bridge_call() and ast_bridge_call_with_flags(), changed the
calling contract to require the initial COLP exchange to already have been
done by the caller.

* Made all callers of ast_bridge_impart() check the return value.  It is
important.  As a precaution, I also made the compiler complain now if it
is not checked.

* Did some cleanup in parking_tests.c as a result of checking the
ast_bridge_impart() return value.

An independent, but associated change is:
* Reduce stack usage in ast_indicate_data() and add a dropping redundant
connected line verbose message.

(closes issue ASTERISK-22072)
Reported by: Joshua Colp

Review: https://reviewboard.asterisk.org/r/2845/
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7 years agoDon't write to /tmp/refs when REF_DEBUG is not defined.
David M. Lee [Fri, 13 Sep 2013 20:55:09 +0000 (20:55 +0000)]
Don't write to /tmp/refs when REF_DEBUG is not defined.

If MALLOC_DEBUG is enabled, then the debug destructor for the container
is used, which would erroneously write to /tmp/refs. This patch only
uses the debug destructor if ref_debug is used.

(closes issue ASTERISK-22536)
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7 years agoCreate more accurate Contact headers for dialogs when we are the UAS.
Mark Michelson [Fri, 13 Sep 2013 14:50:38 +0000 (14:50 +0000)]
Create more accurate Contact headers for dialogs when we are the UAS.

(closes issue AST-1207)
reported by John Bigelow

Review: https://reviewboard.asterisk.org/r/2842
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7 years agoChange how realms are handled for outbound authentication.
Mark Michelson [Fri, 13 Sep 2013 14:44:43 +0000 (14:44 +0000)]
Change how realms are handled for outbound authentication.

With this change, if no realm is specified in an outbound auth
section, then we will simply match the realm that was present
in the 401/407 challenge.

(closes issue ASTERISK-22471)
Reported by George Joseph
(closes issue ASTERISK-22386)
Reported by Rusty Newton

Patches:
outbound_auth_realm_v4.patch uploaded by George Joseph (License #6322)
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7 years agoRecorded merge of revisions 399035,399049 from http://svn.asterisk.org/svn/asterisk...
David M. Lee [Fri, 13 Sep 2013 14:43:56 +0000 (14:43 +0000)]
Recorded merge of revisions 399035,399049 from svn.asterisk.org/svn/asterisk/branches/12

These were lost in r399071

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399081 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoPut merge tracking for r399039 back.
David M. Lee [Fri, 13 Sep 2013 14:34:43 +0000 (14:34 +0000)]
Put merge tracking for r399039 back.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399080 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoBroke the build! Forgot para tags within my description.
Rusty Newton [Fri, 13 Sep 2013 14:27:54 +0000 (14:27 +0000)]
Broke the build! Forgot para tags within my description.

https://bamboo.asterisk.org/bamboo/browse/AST-ATRUNKBUILD-304
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7 years agores_pjsip: Forward PJSIP logging to Asterisk logging
David M. Lee [Fri, 13 Sep 2013 14:22:07 +0000 (14:22 +0000)]
res_pjsip: Forward PJSIP logging to Asterisk logging

This patch uses PJSIP's pj_log_set_log_func() to forward PJSIP's log
messages to Asterisk's logger. This is done in a new module:
res_pjsip_log_forwarder.so.

This patch sets defaultenabled on the existing res_pjsip_logger.so to
no, since logging every SIP packet seems a bit odd to do by default, and
is (hopefully) less necessary with regular PJSIP logging.

It also removes res_rtp_asterisk's disabling of PJSIP logging.

(closes issue ASTERISK-22360)
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/2830/
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7 years agoARI: Fix WebSocket response when subprotocol isn't specified
David M. Lee [Fri, 13 Sep 2013 14:19:19 +0000 (14:19 +0000)]
ARI: Fix WebSocket response when subprotocol isn't specified

When I moved the ARI WebSocket from /ws to /ari/events, I added code to
allow a WebSocket to connect without specifying the subprotocol if
there's only one subprotocol handler registered for the WebSocket.

Naively, I coded it to always respond with the subprotocol in use.
Unfortunately, according to RFC 6455, if the server's response includes
a subprotocol header field that "indicates the use of a subprotocol that
was not present in the client's handshake [...], the client MUST _Fail
the WebSocket Connection_.", emphasis theirs.

This patch correctly omits the Sec-WebSocket-Protocol if one is not
specified by the client.

(closes issue ASTERISK-22441)
Review: https://reviewboard.asterisk.org/r/2828/
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7 years agoFix several crashes in MeetMeAdmin
Kinsey Moore [Fri, 13 Sep 2013 14:17:15 +0000 (14:17 +0000)]
Fix several crashes in MeetMeAdmin

This change ensures that MeetMeAdmin commands requiring a user actually
get a user and fixes another issue where an extra dereference could
occur for a last-entered user being ejected if a user identifier was
also provided.

(closes issue ASTERISK-21907)
Reported by: Alex Epshteyn
Review: https://reviewboard.asterisk.org/r/2844/
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7 years ago'identify' configObject doesn't have a synopsis
Rusty Newton [Fri, 13 Sep 2013 13:28:26 +0000 (13:28 +0000)]
'identify' configObject doesn't have a synopsis

Add a straightforward synopsis and description to the identify config object
in XML documentation.

(issue ASTERISK-22311)
(closes issue ASTERISK-22311)
Reported By: Rusty Newton
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7 years agoCLI bridge: Fix "bridge destroy <id>" and "bridge kick <id> <chan>" tab completion.
Richard Mudgett [Thu, 12 Sep 2013 23:42:23 +0000 (23:42 +0000)]
CLI bridge: Fix "bridge destroy <id>" and "bridge kick <id> <chan>" tab completion.

These two commands must deal with the live bridges container for tab
completion and not the stasis cache.
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7 years agoastobj2: Register the bridges container for debug inspection.
Richard Mudgett [Thu, 12 Sep 2013 23:36:33 +0000 (23:36 +0000)]
astobj2: Register the bridges container for debug inspection.
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7 years agoDocumentation fix and improvements to XML configuration help res_pjsip_acl
Rusty Newton [Thu, 12 Sep 2013 23:23:12 +0000 (23:23 +0000)]
Documentation fix and improvements to XML configuration help res_pjsip_acl

 *  One bug fix. Made the synopsis for "type" to accurate.
 *  changing the usage of "IP-domains" to "IP addresses"
 *  clarifying the usage for the options, by adding a relevant description for
    each
 *  modified other areas of the XML help for clarity, such as the module
    description and a few synopsis changes here and there. See the patch.

(issue ASTERISK-22458)
(closes issue ASTERISK-22458)
Reported By: Rusty Newton
Review: https://reviewboard.asterisk.org/r/2823/
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7 years agochan_sip: Revert r398835 due to failing tests involving originate
Jonathan Rose [Thu, 12 Sep 2013 20:27:56 +0000 (20:27 +0000)]
chan_sip: Revert r398835 due to failing tests involving originate

(issue ASTERISK-22424)
Reported by: Jonathan Rose
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7 years agocore_local: Fix memory corruption race condition.
Richard Mudgett [Thu, 12 Sep 2013 16:44:34 +0000 (16:44 +0000)]
core_local: Fix memory corruption race condition.

The masquerade super test is failing on v12 with high fence violations and
crashing.  The fence violations are showing that party id allocated memory
strings are somehow getting corrupted in the
bridge_reconfigured_connected_line_update() function.  The invalid string
values happen to be the freed memory fill pattern.

After much puzzling, I deduced that the
bridge_reconfigured_connected_line_update() is copying a string out of the
source channel's caller party id struct just as another thread is updating
it with a new value.  The copying thread is using the old string pointer
being freed by the updating thread.  A search of the code found the
unreal_colp_redirect_indicate() routine updating the caller party id's
without holding the channel lock.

A latent bug in v1.8 and v11 hatched in v12 because of the bridging and
connected line changes.  :)

(issue ASTERISK-22221)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/2839/
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7 years agoFix symbol collision with pjsua.
David M. Lee [Thu, 12 Sep 2013 15:23:54 +0000 (15:23 +0000)]
Fix symbol collision with pjsua.

We shouldn't be exporting any symbols that start with pjsip_.
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7 years ago'queue add member' help text correction
Rusty Newton [Thu, 12 Sep 2013 00:04:57 +0000 (00:04 +0000)]
'queue add member' help text correction

You are adding dial strings to the queue, not channels. An aribitrary string
could be used, but you are typically referencing a channel. Correcting the
command help text.

(issue ASTERISK-22263)
(closes issue ASTERISK-22263)
Reported By: Rusty Newton
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7 years agoDocumentation fix - waitfordialtone is not boolean, it's time in milliseconds
Rusty Newton [Wed, 11 Sep 2013 23:52:49 +0000 (23:52 +0000)]
Documentation fix - waitfordialtone is not boolean, it's time in milliseconds

Changing text in chan_dahdi.conf sample to be accurate.

(issue ASTERISK-22308)
(closes issue ASTERISK-22308)
Reported By: Malcolm Davenport
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7 years agochan_sip: Reject calls without prior SDP on 200 OK
Jonathan Rose [Wed, 11 Sep 2013 20:03:19 +0000 (20:03 +0000)]
chan_sip: Reject calls without prior SDP on 200 OK

If we receive a 200 OK without SDP, we will now check to see if
the remote address has been established for that channel's RTP
session and if the to tag for that channel has changed from
the most recent to tag in a response less than 200.
If either a change has been made since the last to-tag was
received or the remote address is unset, then we will drop
the call.

(closes issue ASTERISK-22424)
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/2827/diff/#index_header
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7 years agoFix typo in confbridge.conf.sample
Russell Bryant [Wed, 11 Sep 2013 18:03:30 +0000 (18:03 +0000)]
Fix typo in confbridge.conf.sample

The denoise filter requires func_speex, not codec_speex.  Fix this in the
description of the denoise=yes option in confbridge.conf.
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7 years agopjsip: reinvite for connected line updates occurs when it should not
Kevin Harwell [Wed, 11 Sep 2013 14:23:28 +0000 (14:23 +0000)]
pjsip: reinvite for connected line updates occurs when it should not

Connected line updates are now only sent out if an actual update needs to occur.
This happens under the following conditions:

1. The endpoint we are sending to is trusted.
2. Either a P-Asserted-Identity or Remote Party-ID header needs to be added/sent.
3. The connected id's number and name are valid.

Also added an SDP when an update is sent out.

(closes issue AST-1212)
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/2831/
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7 years agoFix incorrect usages of ast_realloc().
Richard Mudgett [Tue, 10 Sep 2013 18:05:47 +0000 (18:05 +0000)]
Fix incorrect usages of ast_realloc().

There are several locations in the code base where this is done:
buf = ast_realloc(buf, new_size);

This is going to leak the original buf contents if the realloc fails.

Review: https://reviewboard.asterisk.org/r/2832/
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7 years agoFixed utils directory breakage from r398748, this time with extra hate.
David M. Lee [Tue, 10 Sep 2013 17:50:13 +0000 (17:50 +0000)]
Fixed utils directory breakage from r398748, this time with extra hate.
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7 years agoFixed utils directory breakage from r398648
David M. Lee [Tue, 10 Sep 2013 17:26:19 +0000 (17:26 +0000)]
Fixed utils directory breakage from r398648
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7 years agoMALLOC_DEBUG: Change fence magic number to be completely different from the freed...
Richard Mudgett [Mon, 9 Sep 2013 23:29:44 +0000 (23:29 +0000)]
MALLOC_DEBUG: Change fence magic number to be completely different from the freed magic number.

Race conditions between freeing a nul terminated string and
ast_strdup()'ing it are more likely to be detected if the fence and freed
magic numbers are completely different.
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7 years agoAdd extra debugging to res_pjsip_endpoint_identifier_ip
Mark Michelson [Mon, 9 Sep 2013 22:00:44 +0000 (22:00 +0000)]
Add extra debugging to res_pjsip_endpoint_identifier_ip
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Merged revisions 398694 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398695 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoFix DEBUG_THREADS when lock is acquired in __constructor__
David M. Lee [Mon, 9 Sep 2013 20:13:40 +0000 (20:13 +0000)]
Fix DEBUG_THREADS when lock is acquired in __constructor__

This patch fixes some long-standing bugs in debug threads that were
exacerbated with recent Optional API work in Asterisk 12.

With debug threads enabled, on some systems, there's a lock ordering
problem between our mutex and glibc's mutex protecting its module list
(Ubuntu Lucid, glibc 2.11.1 in this instance). In one thread, the module
list will be locked before acquiring our mutex. In another thread, our
mutex will be locked before locking the module list (which happens in
the depths of calling backtrace()).

This patch fixes this issue by moving backtrace() calls outside of
critical sections that have the mutex acquired. The bigger change was to
reentrancy tracking for ast_cond_{timed,}wait, which wrongly assumed
that waiting on the mutex was equivalent to a single unlock (it actually
suspends all recursive locks on the mutex).

(closes issue ASTERISK-22455)
Review: https://reviewboard.asterisk.org/r/2824/
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Merged revisions 398648 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 398651 from http://svn.asterisk.org/svn/asterisk/branches/12

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7 years agoMultiple revisions 398638-398639
David M. Lee [Mon, 9 Sep 2013 19:09:21 +0000 (19:09 +0000)]
Multiple revisions 398638-398639

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  r398638 | dlee | 2013-09-09 14:01:54 -0500 (Mon, 09 Sep 2013) | 1 line

  Added note about expected behavior of originate
........
  r398639 | dlee | 2013-09-09 14:02:27 -0500 (Mon, 09 Sep 2013) | 1 line

  Added note about expected behavior of originate (the rest of the commit)
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7 years agoBlocked revisions 398559,398578
David M. Lee [Mon, 9 Sep 2013 19:08:15 +0000 (19:08 +0000)]
Blocked revisions 398559,398578

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Multiple revisions 398559,398578

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  r398559 | kmoore | 2013-09-06 14:32:03 -0500 (Fri, 06 Sep 2013) | 20 lines

  Blocked revisions 398558

  ........
  Fix Jabber/XMPP distributed MWI

  The mailbox and context are swapped on the receiving end for all users
  of Jabber and XMPP distributed MWI in Asterisk 1.8 and all more recent
  versions. This swaps those values to be correct when publishing to the
  internal event system from Jabber/XMPP distributed MWI state.

  (closes issue ASTERISK-22435)
  Reported by: abelbeck
  Tested by: Michael Keuter
  Patches:
      asterisk-1.8-res_jabber-aji_handle_pubsub_event.patch uploaded by abelbeck
      asterisk-11-res_xmpp-xmpp_pubsub_handle_event.patch uploaded by abelbeck
  ........

  Merged revisions 398523 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
  r398578 | kmoore | 2013-09-06 16:03:45 -0500 (Fri, 06 Sep 2013) | 1 line

  Unblock r398558

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398640 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoUpdate CDR Unit tests to reflect container changes in r398579
Matthew Jordan [Sun, 8 Sep 2013 23:30:39 +0000 (23:30 +0000)]
Update CDR Unit tests to reflect container changes in r398579

When a channel joins a multi-party bridge, the ordering of the CDRs that is
created is determined by the ordering of the channels who happen to be in that
bridge. When r398579 changed the number of buckets in the container to
something sensible, it changed the ordering that the CDRs was created in,
causing one of the multiparty tests to fail. This fixes the test with the
now expected ordering.
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Merged revisions 398628 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398629 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoPrevent XMPP timeout on blank responses
Kinsey Moore [Sat, 7 Sep 2013 01:03:07 +0000 (01:03 +0000)]
Prevent XMPP timeout on blank responses

Sometimes the Google Voice servers have a bad habit of sending out 1
byte replies to the xmpp resource. When a blank 1 byte reply is
received from the socket the buffer attempts to wait (endlessly) for
the rest of the reply from google which effectively blocks the socket
and google voice calls will no longer come into the server.

This patch allows the xmpp module to correctly detect empty packets and
send out ping replies to google. It also sets a socket timeout on the
default socket which prevents the xmpp socket from closing and
preventing future google voice calls from coming into the server.

Furthermore instead of sending an empty reply back to google we send a
proper xmpp ping reply back. This also adds several more
socket messages.

(closes issue ASTERISK-22347)
Reported by: Andrew Nagy
Review: https://reviewboard.asterisk.org/r/2771
Patches:
    xmpp_fix_1.diff uploaded by Andrew Nagy (License #6524)
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Merged revisions 398618 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 398619 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398620 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoMultiple revisions 398558,398577
Kinsey Moore [Fri, 6 Sep 2013 21:23:02 +0000 (21:23 +0000)]
Multiple revisions 398558,398577

........
  r398558 | kmoore | 2013-09-06 14:28:16 -0500 (Fri, 06 Sep 2013) | 17 lines

  Fix Jabber/XMPP distributed MWI

  The mailbox and context are swapped on the receiving end for all users
  of Jabber and XMPP distributed MWI in Asterisk 1.8 and all more recent
  versions. This swaps those values to be correct when publishing to the
  internal event system from Jabber/XMPP distributed MWI state.

  (closes issue ASTERISK-22435)
  Reported by: abelbeck
  Tested by: Michael Keuter
  Patches:
      asterisk-1.8-res_jabber-aji_handle_pubsub_event.patch uploaded by abelbeck
      asterisk-11-res_xmpp-xmpp_pubsub_handle_event.patch uploaded by abelbeck
  ........

  Merged revisions 398523 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
  r398577 | kmoore | 2013-09-06 16:00:56 -0500 (Fri, 06 Sep 2013) | 10 lines

  Commit the remainder of r398523

  This is a missing part of the commit in revision 398523 that corrects
  the name of a variable.

  (issue ASTERISK-22435)
  ........

  Merged revisions 398576 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 398558,398577 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 398580 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398603 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agocdr: Change the number of container buckets to be similar to the channels container.
Richard Mudgett [Fri, 6 Sep 2013 21:17:45 +0000 (21:17 +0000)]
cdr: Change the number of container buckets to be similar to the channels container.

* Fix the temporary cdr candidate containers to use a prime number of
buckets.
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Merged revisions 398579 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398583 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agocore_local: Fix LocalOptimizationBegin AMI event missing Source channel snapshot.
Richard Mudgett [Fri, 6 Sep 2013 20:21:21 +0000 (20:21 +0000)]
core_local: Fix LocalOptimizationBegin AMI event missing Source channel snapshot.

* Fix the LocalOptimizationBegin AMI event by eliminating an artificial
buffer size limitation that is too small anyway.
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Merged revisions 398572 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398574 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agocdr: Fix some ref leaks.
Richard Mudgett [Fri, 6 Sep 2013 20:03:01 +0000 (20:03 +0000)]
cdr: Fix some ref leaks.

* Added missing unregister of the cdr container in cdr_engine_shutdown().

* Fixed ref leak in off nominal path of cdr_object_alloc().

* Removed some unnecessary NULL checks in cdr_object_dtor().
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Merged revisions 398562 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398564 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoastobj2: Add warn unused attribute to some functions.
Richard Mudgett [Fri, 6 Sep 2013 19:26:48 +0000 (19:26 +0000)]
astobj2: Add warn unused attribute to some functions.

* Fixed resulting warnings with improper use of ao2_global_obj_replace().

* Made a couple uses of ao2_global_obj_replace_unref(x, NULL) into the
equivalent and more appropriate ao2_global_obj_release() call.
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Merged revisions 398533 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398557 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoFix build warnings
Kinsey Moore [Fri, 6 Sep 2013 18:53:32 +0000 (18:53 +0000)]
Fix build warnings

When AST_DEVMODE is not defined, ast_asserts are not compiled into the
binary. In some cases, this means variables are not referenced or are
set but unused which causes warnings to show up.

(closes issue ASTERISK-22446)
Reported by: Jason Parker (qwell)
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Merged revisions 398521 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398522 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoFix chan_h323 compilation
Kinsey Moore [Fri, 6 Sep 2013 16:01:05 +0000 (16:01 +0000)]
Fix chan_h323 compilation

This fixes the things in chan_h323 that were missed or ignored in the
great channel opaquification and gets chan_h323 back into a compiling
state.

(closes issue ASTERISK-22365)
Reported by: Dmitry Melekhov
Patches:
    chan_h323.patch uploaded by Dmitry Melekhov
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Merged revisions 398510 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 398511 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398512 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoastobj2: Only define ao2_bt() once.
Richard Mudgett [Thu, 5 Sep 2013 21:48:02 +0000 (21:48 +0000)]
astobj2: Only define ao2_bt() once.

* Make ao2_bt() not use single char variable names.

* Fix ao2_bt() formatting.
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Merged revisions 398498 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398499 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agochan_iax2: Reduce indentation in __attempt_transmit().
Richard Mudgett [Thu, 5 Sep 2013 19:18:10 +0000 (19:18 +0000)]
chan_iax2: Reduce indentation in __attempt_transmit().

* Reduce indentation in __attempt_transmit().

* Don't update the static last error time variable every time in
__schedule_action() and socket_read().
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Merged revisions 398456 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 398457 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 398458 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398462 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agochan_iax2: Fix stray reference to worker thread idle_list.
Richard Mudgett [Thu, 5 Sep 2013 17:31:29 +0000 (17:31 +0000)]
chan_iax2: Fix stray reference to worker thread idle_list.

* Fix stray reference to idle_list in cleanup_thread_list().  This may be
the reason for the note in iax2_process_thread() about threads not being
removed from the task lists.

* Move cleanup_thread_list(&idle_list) to after the other lists are
cleaned up.
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Merged revisions 398416 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 398417 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 398418 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398419 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agochan_iax2: Fix bridgecallno deadlock avoidance.
Richard Mudgett [Thu, 5 Sep 2013 17:17:53 +0000 (17:17 +0000)]
chan_iax2: Fix bridgecallno deadlock avoidance.

* Fix bridgecallno deadlock avoidance.  When doing deadlock avoidance, you
need to retest the status of values for each loop to see if you still need
the lock for bridgecallno.

* As a safety check, after acquiring the bridgecallno lock you should
check if iaxs[bridgecallno] is NULL just like the current callno checks.

* Move setting thread->iostate to IAX_IOSTATE_IDLE to after processing any
deferred frames to ensure that the iostate is IDLE when it is placed back
into the idle list.  defer_full_frame() tries to ensure
iax2_process_thread() wakes up to process the frame.
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Merged revisions 398379 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 398380 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 398381 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398384 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoClarify server_uri and client_uri registration settings.
Mark Michelson [Thu, 5 Sep 2013 14:10:45 +0000 (14:10 +0000)]
Clarify server_uri and client_uri registration settings.

Used some of Rusty's suggested language plus also included
more SIPesque descriptions of where the URIs are actually
used in an outgoing REGISTER.

(closes issue ASTERISK-22390)
reported by Rusty Newton
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Merged revisions 398368 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398369 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agochan_iax2: Add missing control frame names to debug frame decode output.
Richard Mudgett [Wed, 4 Sep 2013 23:07:41 +0000 (23:07 +0000)]
chan_iax2: Add missing control frame names to debug frame decode output.
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Merged revisions 398301 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 398302 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 398303 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398304 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoGive more detail regarding failures to create request with auth credentials.
Mark Michelson [Wed, 4 Sep 2013 22:49:25 +0000 (22:49 +0000)]
Give more detail regarding failures to create request with auth credentials.

(issue ASTERISK-22386)
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Merged revisions 398299 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398300 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agounit tests: test_voicemail_api leaks stringfields from snapshots
Jonathan Rose [Wed, 4 Sep 2013 21:37:01 +0000 (21:37 +0000)]
unit tests: test_voicemail_api leaks stringfields from snapshots

(closes issue ASTERISK-22414)
Reported by: Corey Farrell
Patches:
    test_voicemail_api-leaks-11.patch uploaded by coreyfarrell (license 5909)
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Merged revisions 398285 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 398286 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398287 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoapp_voicemail: Fix leaking config objects when msg_id doesn't match
Jonathan Rose [Wed, 4 Sep 2013 21:20:02 +0000 (21:20 +0000)]
app_voicemail: Fix leaking config objects when msg_id doesn't match

(issues ASTERISK-22414)
Reported by: Corey Farrell
Patch:
    test_voicemail_api-leaks-11.patch uploaded by coreyfarrell (license 5909)
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Merged revisions 398281 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 398283 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398284 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agochan_misdn: Fix misdn debug output printed with arbitrary verbose levels.
Richard Mudgett [Wed, 4 Sep 2013 16:03:14 +0000 (16:03 +0000)]
chan_misdn: Fix misdn debug output printed with arbitrary verbose levels.

Fix the misdn debug output to remote consoles.  chan_misdn uses
ast_console_puts() which doesn't know about verbose levels.  Better to use
ast_verbose() instead.  Without this patch the misdn debug messages are
appended to the verbose level which ever was set by the message sent to
the console before, i.e.  any undefined level.

(closes issue AST-1218)
Reported by: Guenther Kelleter
Patches:
      misdnlog.patch (license #6372) patch uploaded by Guenther Kelleter
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Merged revisions 398235 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 398236 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 398237 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398238 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoDebug messages for pjsip outbound registration
Kevin Harwell [Wed, 4 Sep 2013 14:32:25 +0000 (14:32 +0000)]
Debug messages for pjsip outbound registration

Added debug messages indicating that an outbound registration attempt was made
and it was successful in pjsip.

(closes issue ASTERISK-22388)
Reported by: Rusty Newton
........

Merged revisions 398226 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398227 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoFix remote tcs sequence handling on empty tcs received
Alexandr Anikin [Tue, 3 Sep 2013 20:28:01 +0000 (20:28 +0000)]
Fix remote tcs sequence handling on empty tcs received
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Merged revisions 398214 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 398215 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398217 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoPrevent a crash in res_pjsip_dtmf_info.c
Kinsey Moore [Tue, 3 Sep 2013 18:09:02 +0000 (18:09 +0000)]
Prevent a crash in res_pjsip_dtmf_info.c

This change makes sure that a content type header exists before
checking the contents of the header against known SIP INFO DTMF content
types.
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Merged revisions 398206 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398207 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoFixed 'make clean' for wiki docs
David M. Lee [Tue, 3 Sep 2013 17:19:30 +0000 (17:19 +0000)]
Fixed 'make clean' for wiki docs
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Merged revisions 398198 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398205 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoBe a little more verbose when loading cel_custom.conf.
Walter Doekes [Tue, 3 Sep 2013 14:29:52 +0000 (14:29 +0000)]
Be a little more verbose when loading cel_custom.conf.

Review: https://reviewboard.asterisk.org/r/2805/
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Merged revisions 398167 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 398168 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 398196 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398197 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoFix graceful shutdown crash.
David M. Lee [Fri, 30 Aug 2013 20:58:59 +0000 (20:58 +0000)]
Fix graceful shutdown crash.

The cleanup code for optional_api needs to happen after all of the optional
API users and providers have unused/unprovided. Unfortunately, regsitering the
atexit() handler at the beginning of main() isn't soon enough, since module
destructors run after that.
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Merged revisions 398149 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398150 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoNew pjsip.conf.sample
Rusty Newton [Fri, 30 Aug 2013 20:37:54 +0000 (20:37 +0000)]
New pjsip.conf.sample

(issue ASTERISK-22145)
(closes issue ASTERISK-22145)
Reported By: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2811/
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Merged revisions 398147 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398148 65c4cc65-6c06-0410-ace0-fbb531ad65f3