asterisk/asterisk.git
5 years agores_stasis: Extend bridge type to be a comma separated list of bridge attributes.
Joshua Colp [Wed, 19 Mar 2014 12:54:25 +0000 (12:54 +0000)]
res_stasis: Extend bridge type to be a comma separated list of bridge attributes.

This change turns the bridge type field into a comma separated list of attributes.
These attributes include: mixing, holding, dtmf_events, and proxy_media. By setting
the various attributes a user can control the type of bridge created with the
behavior they need for their application.

(closes issue ASTERISK-23437)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/3359/
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Merged revisions 410904 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410905 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agores_ari: Fix documentation schema error
Matthew Jordan [Wed, 19 Mar 2014 02:33:55 +0000 (02:33 +0000)]
res_ari: Fix documentation schema error
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Merged revisions 410890 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410891 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agores_ari: Add notes about Asterisk HTTP server to the "enabled" config option for...
Rusty Newton [Tue, 18 Mar 2014 23:32:00 +0000 (23:32 +0000)]
res_ari: Add notes about Asterisk HTTP server to the "enabled" config option for the res_ari general section

Added note and see-also reminding user to enable the HTTP server.

(closes issue ASTERISK-22499)
Reported by: Rusty Newton
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Merged revisions 410876 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410877 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoARI: allow json content type with zero length body
Scott Griepentrog [Tue, 18 Mar 2014 15:45:04 +0000 (15:45 +0000)]
ARI: allow json content type with zero length body

When a request was received with a Content-type of json,
the body was sent for json parsing - even if it was zero
length.  This resulted in ARI requests failing that were
valid, such as a channel DELETE with no parameters.  The
code has now been changed to skip json parsing with zero
content length.

(closes issue SWP-6748)
Reported by: Samuel Galarneau
Review: https://reviewboard.asterisk.org/r/3360/
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Merged revisions 410858 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410863 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agocdr: Add asserts for when we don't know about a CDR for a channel
Matthew Jordan [Tue, 18 Mar 2014 15:28:45 +0000 (15:28 +0000)]
cdr: Add asserts for when we don't know about a CDR for a channel

In the CDR core, every channel should either be filtered out (due to being an
'internal' channel used as an implementation detail, such as playing media
back into a bridge) or it should get a CDR. Even if that CDR ends up being
discarded, we still give the channel a CDR in case we end up needing it. If we
hit a situation where a channel does not have a CDR, we should blow up in
-dev-mode. Asserts are appropriate for that.

This patch adds those asserts, as they would have quickly caught the error
fixed by r410814.
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Merged revisions 410861 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410862 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agores_pjsip: Fix memory leak of nameservers in off-nominal resolver creation failure.
Joshua Colp [Tue, 18 Mar 2014 12:45:49 +0000 (12:45 +0000)]
res_pjsip: Fix memory leak of nameservers in off-nominal resolver creation failure.

Thanks Walter Doekes!
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Merged revisions 410844 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410845 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agores_fax_spandsp: Use g711_free() when available.
Sean Bright [Tue, 18 Mar 2014 11:52:15 +0000 (11:52 +0000)]
res_fax_spandsp: Use g711_free() when available.

Per Johann Steinwendtner on the asterisk-dev mailing list:

http://lists.digium.com/pipermail/asterisk-dev/2014-March/066102.html

g711_free() was introduced in spandsp 0.0.6pre4 and g711_release() became a
noop.  I opted not to remove the call to g711_release() since it is harmless
and to call g711_free() if we have a sufficiently recent version of spandsp.

(issue ASTERISK-20149)
Reported by: Alexandr Gordeev
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Merged revisions 410829 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 410830 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410831 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agostasis_cache: Use the right variable in the cache entry ao2 cmp function.
Richard Mudgett [Tue, 18 Mar 2014 02:09:25 +0000 (02:09 +0000)]
stasis_cache: Use the right variable in the cache entry ao2 cmp function.
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Merged revisions 410813 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410814 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agores_pjsip: Enable PJSIP DNS client support.
Joshua Colp [Mon, 17 Mar 2014 22:54:32 +0000 (22:54 +0000)]
res_pjsip: Enable PJSIP DNS client support.

This change enables DNS client support within PJSIP. System
nameservers are automatically discovered using res_init or
res_ninit. If this fails then PJSIP will resort to using
gethostbyname for resolution.

By enabling this support we gain SRV support, failover, and
weight support.

(closes issue ASTERISK-23435)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/3343/
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Merged revisions 410795 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410796 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agores_pjsip_multihomed: Make address replacement less aggressive.
Joshua Colp [Mon, 17 Mar 2014 22:46:56 +0000 (22:46 +0000)]
res_pjsip_multihomed: Make address replacement less aggressive.

This change makes the res_pjsip_multihomed module less aggressive when
changing the address in messages. It will now only occur if the transport
in use is bound to the any address OR if the system determined source
address matches the bound address of the transport in use.

Review: https://reviewboard.asterisk.org/r/3369/
........

Merged revisions 410793 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410794 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agocallerid: Logic error in checksum processing
Russ Meyerriecks [Mon, 17 Mar 2014 22:24:03 +0000 (22:24 +0000)]
callerid: Logic error in checksum processing

Callerid checksum-ing was being handled incorrectly here. When the checksum is
calculated to be 0x00, it will perform 0x100-0x00 which results in 0x100. This
value will then fail the otherwise correct callerid message.

This patch changes the logic to simply add the calculated checksum to the
transmitted 2's compliment checksum.

Review: https://reviewboard.asterisk.org/r/3356/
(closes issue ASTERISK-23488)

........
This is a merge of merged revisions 410750 410747 from http://svn.asterisk.org/svn/asterisk/branches/12
I didn't want a broken patch to be comitted to trunk so I pre-merge merged them.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410775 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoRevert changes to sorcery that accidentally got committed.
Mark Michelson [Mon, 17 Mar 2014 19:35:17 +0000 (19:35 +0000)]
Revert changes to sorcery that accidentally got committed.

These changes were still up for review and have not been approved
yet. I must have had the changes in my working copy when making
a different change.
........

Merged revisions 410696 from http://svn.asterisk.org/svn/asterisk/branches/12

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5 years agoFix stuck channel in ARI through the introduction of synchronous bridge actions.
Mark Michelson [Mon, 17 Mar 2014 17:22:12 +0000 (17:22 +0000)]
Fix stuck channel in ARI through the introduction of synchronous bridge actions.

Playing back a file to a channel in an ARI bridge would attempt to wait until
the playback concluded before returning. The method used involved signaling the
waiting thread in the ARI custom playback function.

The problem with this is that there were some corner cases that were not accounted for:
* If a bridge channel could not be found, then we never would attempt the playback but
  would still attempt to wait for the playback to complete.
* If the bridge playfile action failed to queue, we would still attempt to wait for the
  playback to complete.
* If the bridge playfile action were queued but some circumstance caused the playback
  not to occur (the bridge dies, the channel is removed from the bridge), then we would
  never be notified.

The solution to this is to move the waiting logic into the bridge code. A new bridge
API function is added to queue a synchronous action on a bridge. The waiting thread
is notified when the queued frame has been freed, either due to an error occurring
or due to successful playback. As a failsafe, the waiting thread has a 10 minute
timeout just in case there is a frame leak somewhere.

Review: https://reviewboard.asterisk.org/r/3338
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Merged revisions 410673 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410684 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoapp_confbridge: Add missing destructor call to announcer channel destructor.
Richard Mudgett [Mon, 17 Mar 2014 16:48:55 +0000 (16:48 +0000)]
app_confbridge: Add missing destructor call to announcer channel destructor.
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Merged revisions 410671 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410672 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agostasis/app.c: Add some extra debugging for subscription counts
Matthew Jordan [Sun, 16 Mar 2014 20:27:28 +0000 (20:27 +0000)]
stasis/app.c: Add some extra debugging for subscription counts

Events are sent to a connected ARI application based on the things that ARI
application cares about. These subscriptions can be set up implicitly - such
as when that ARI application creates a new object - or explicitly, via the
application resource's subscription operations. Debugging *why* something was
being sent to an application - or why something was not being sent to an
application - was a bit tricky, as there was no debug information for the
subscriptions.

This patch adds some debug level 3 statements that show the subscription counts
for applications. (Level 3 was chosen as it matches the verbose level 3
statements elsewhere)
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Merged revisions 410650 from http://svn.asterisk.org/svn/asterisk/branches/12

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5 years agoframehook.h: Fix some doc typos.
Russell Bryant [Sat, 15 Mar 2014 15:24:23 +0000 (15:24 +0000)]
framehook.h: Fix some doc typos.

There were a number of instances in this header file where "function all" was
intended to be "function call".  This patch fixes that up.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410639 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoFix failing realtime sorcery tests.
Mark Michelson [Fri, 14 Mar 2014 21:56:21 +0000 (21:56 +0000)]
Fix failing realtime sorcery tests.

The store realtime callback needs to return a positive value for
sorcery to treat the store as a success.
........

Merged revisions 410625 from http://svn.asterisk.org/svn/asterisk/branches/12

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5 years agomanager: fix memory leak in manager_add_filter function
Jonathan Rose [Fri, 14 Mar 2014 21:36:55 +0000 (21:36 +0000)]
manager: fix memory leak in manager_add_filter function

(closes issue ASTERISK-23420)
Reported by: Etienne Lessard
Patches:
    manager_eventfilter_leak uploaded by Etienne Lessard (license 6394)
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Merged revisions 410609 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 410623 from http://svn.asterisk.org/svn/asterisk/branches/12

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5 years agoRemove an extra ast_cond_wait() that slipped through the patch.
Mark Michelson [Fri, 14 Mar 2014 20:55:06 +0000 (20:55 +0000)]
Remove an extra ast_cond_wait() that slipped through the patch.
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Merged revisions 410606 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 410607 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410608 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoHandle the return values of realtime updates and stores more accurately.
Mark Michelson [Fri, 14 Mar 2014 18:11:55 +0000 (18:11 +0000)]
Handle the return values of realtime updates and stores more accurately.

Realtime backends' update and store callbacks return the number of rows affected,
or -1 if there was a failure. There were a couple of issues:

* The config API was treating 0 as a successful return, and positive values as
  a failure. Now the config API treats anything >= 0 as a success.

* res_sorcery_realtime was treating 0 as a successful return from the store
  procedure, and any positive values as a failure. Now sorcery treats anything
  > 0 as a success. It still considers 0 a "failure" since there is no change
  to report to observers.

Review: https://reviewboard.asterisk.org/r/3341
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Merged revisions 410592 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410593 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoPrevent conflicts regarding unsolicited and solicited MWI to an endpoint.
Mark Michelson [Fri, 14 Mar 2014 18:05:04 +0000 (18:05 +0000)]
Prevent conflicts regarding unsolicited and solicited MWI to an endpoint.

If an endpoint is receiving unsolicited MWI for a mailbox and then attempts
to subscribe to an AOR that provides MWI for the same mailbox, then the SUBSCRIBE
is rejected with a 500 response.

Review: https://reviewboard.asterisk.org/r/3345
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Merged revisions 410590 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410591 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agouniqueid: Update CHANGES to reflect new features
Scott Griepentrog [Fri, 14 Mar 2014 17:56:53 +0000 (17:56 +0000)]
uniqueid: Update CHANGES to reflect new features

Note the new features provided by uniqueid in the
CHANGES file.

(issue ASTERISK-23120)
Review: https://reviewboard.asterisk.org/r/3316/
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Merged revisions 410588 from http://svn.asterisk.org/svn/asterisk/branches/12

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5 years agoPJSIP: TOS values should be represented as decimals in sorcery objects
Jonathan Rose [Fri, 14 Mar 2014 16:42:54 +0000 (16:42 +0000)]
PJSIP: TOS values should be represented as decimals in sorcery objects

(closes issue ASTERISK-23235)
Reported by: George Joseph
Review: https://reviewboard.asterisk.org/r/3324/
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Merged revisions 410574 from http://svn.asterisk.org/svn/asterisk/branches/12

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5 years agoPrevent delayed astdb syncs.
Mark Michelson [Fri, 14 Mar 2014 16:19:21 +0000 (16:19 +0000)]
Prevent delayed astdb syncs.

The syncing thread sleeps for a second before waiting to be
told to attempt to sync again. If a signal were sent during this
sleeping period, we would end up having to wait until the next
sync signal occurred in order to sync up the astdb.

This code rearrangement also ensures that any pending transactions
will be synced prior to Asterisk shutting down.

Patches: db_sync.patch by John Hardin (License #6512)
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Merged revisions 410556 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 410559 from http://svn.asterisk.org/svn/asterisk/branches/12

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5 years agoARI/bridges: Forward Playback/Recording Started/Finished to bridge topic
Jonathan Rose [Fri, 14 Mar 2014 16:17:26 +0000 (16:17 +0000)]
ARI/bridges: Forward Playback/Recording Started/Finished to bridge topic

(closes issue ASTERISK-23444)
Reported by: Ben Merrills
Review: https://reviewboard.asterisk.org/r/3340/
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Merged revisions 410558 from http://svn.asterisk.org/svn/asterisk/branches/12

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5 years agores_mwi_external: Clear the stasis cache entry when the external MWI is deleted.
Richard Mudgett [Fri, 14 Mar 2014 16:01:13 +0000 (16:01 +0000)]
res_mwi_external: Clear the stasis cache entry when the external MWI is deleted.

One of the things missing when external MWI support was added was the
ability to clear the stasis cache entry of deleted external MWI mailboxes.

Review: https://reviewboard.asterisk.org/r/3325/
........

Merged revisions 410555 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410557 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agocdr.c: Add missing aow_unlock(cdr) in off nominal path of handle_dial_message().
Richard Mudgett [Thu, 13 Mar 2014 21:27:15 +0000 (21:27 +0000)]
cdr.c: Add missing aow_unlock(cdr) in off nominal path of handle_dial_message().

* Trivial common code hoisting in handle_bridge_leave_message().

* Some whitespace fixing.
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Merged revisions 410541 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410542 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoARI: Ensure managing application receives ChannelEnteredBridge messages
Kinsey Moore [Thu, 13 Mar 2014 19:33:22 +0000 (19:33 +0000)]
ARI: Ensure managing application receives ChannelEnteredBridge messages

This fixes an issue where a Stasis application running over ARI and
subscribed to ari/events could miss the ChannelEnteredBridge event
because it did not subscribe to the new bridge fast enough.

To accomplish this, it subscribes the application controlling the
channel to the new bridge before adding it to that bridge which
required the stasis_app_control structure to maintain a reference to
the stasis_app.

(closes issue ASTERISK-23295)
Review: https://reviewboard.asterisk.org/r/3336/
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Merged revisions 410527 from http://svn.asterisk.org/svn/asterisk/branches/12

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5 years agoMultiple revisions 410509-410510
Joshua Colp [Thu, 13 Mar 2014 13:25:09 +0000 (13:25 +0000)]
Multiple revisions 410509-410510

........
  r410509 | file | 2014-03-13 06:23:14 -0700 (Thu, 13 Mar 2014) | 2 lines

  res_pjsip_multihomed: Fix a bug where the 200 OK for a REGISTER would contain the wrong contact.
........
  r410510 | file | 2014-03-13 06:24:17 -0700 (Thu, 13 Mar 2014) | 2 lines

  res_pjsip_multihomed: Remove change for testing fix.
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5 years agores_musiconhold.c: Generate MOH start/stop events whenever the MOH stream is started...
Richard Mudgett [Wed, 12 Mar 2014 19:06:52 +0000 (19:06 +0000)]
res_musiconhold.c: Generate MOH start/stop events whenever the MOH stream is started/stopped.

* Made res_musiconhold.c always post the MusicOnHoldStart/MusicOnHoldStop
events when it actually starts/stops the music streams.  This allows the
events to always happen when MOH starts/stops.  The event posting code was
moved to the MOH alloc/release routines.

* Made channel_do_masquerade() stop any MOH on the original channel before
masquerading so the original channel will get a stop event with correct
information.

* Cleaned up a couple odd codings in moh_files_alloc() and moh_alloc()
dealing with the music state variable.

(issue ASTERISK-23311)
Reported by: Benjamin Keith Ford

Review: https://reviewboard.asterisk.org/r/3306/
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5 years agoapp_confbridge: Make explicitly stop MOH if a user is kicked or hangs up while MOH...
Richard Mudgett [Wed, 12 Mar 2014 18:47:10 +0000 (18:47 +0000)]
app_confbridge: Make explicitly stop MOH if a user is kicked or hangs up while MOH is playing.

When MOH is playing to a user in a conference and the user is kicked or
hangs up from the conference then the AMI MusicOnHoldStop events didn't
happen.  (Asterisk v11 AMI event: MusicOnHold, state:Stop)

(closes issue ASTERISK-23311)
Reported by: Benjamin Keith Ford

Review: https://reviewboard.asterisk.org/r/3306/
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5 years agores_pjsip_multihomed: Fix a bug where outgoing messages for TCP would go out using...
Joshua Colp [Wed, 12 Mar 2014 12:51:34 +0000 (12:51 +0000)]
res_pjsip_multihomed: Fix a bug where outgoing messages for TCP would go out using UDP.

This change fixes a bug where the code which changes the transport did not check whether
the message is going out over UDP or not before changing it. For TCP and TLS transports
we don't need to change the transport as the correct one is already chosen.
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Merged revisions 410471 from http://svn.asterisk.org/svn/asterisk/branches/12

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5 years agores_pjsip_multihomed: Add module which places the correct address within messages.
Joshua Colp [Tue, 11 Mar 2014 16:07:42 +0000 (16:07 +0000)]
res_pjsip_multihomed: Add module which places the correct address within messages.

Due to how messages are handled within PJSIP it is not until a message is actually
sent that the destination is reliably known. This means that the addresses placed
within the message may not be of the interface the message is being sent out on.

This module determines what interface a message is being sent on and updates the
message to contain the correct address if applicable.

This module was tested by myself in a virtualized environment with multiple interfaces
and also by Kinsey Moore in the following configuration:

Networks:
* 10.24.16.0/21
** hard phone
** default gateway
* 10.24.64.0/21
** softphone with pjsip-based stack

Transport details:
bind address: 0.0.0.0
protocol: UDP

All endpoints were tested with explicitly configured transports and unconfigured transports.

This was tested with inbound and outbound calls, both of which were experiencing detrimental
effects from incorrect IP addresses in SIP messages. These effects were only experienced by the
soft phone on the 10.24.64.0 network since the messages to the hard phone on the 10.24.16.0
network had the correct IP address.

(closes issue ASTERISK-23020)
Reported by: xrobau

Review: https://reviewboard.asterisk.org/r/3102/
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5 years agoAST-2014-001: Stack overflow in HTTP processing of Cookie headers.
Richard Mudgett [Mon, 10 Mar 2014 17:21:01 +0000 (17:21 +0000)]
AST-2014-001: Stack overflow in HTTP processing of Cookie headers.

Sending a HTTP request that is handled by Asterisk with a large number of
Cookie headers could overflow the stack.

Another vulnerability along similar lines is any HTTP request with a
ridiculous number of headers in the request could exhaust system memory.

(closes issue ASTERISK-23340)
Reported by: Lucas Molas, researcher at Programa STIC, Fundacion; and Dr. Manuel Sadosky, Buenos Aires, Argentina
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Merged revisions 410383 from http://svn.asterisk.org/svn/asterisk/branches/12

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5 years agounqiueid: correct max uniqueid length test
Scott Griepentrog [Mon, 10 Mar 2014 16:33:10 +0000 (16:33 +0000)]
unqiueid: correct max uniqueid length test

This patch adds null string test prior to checking for
a max uniqueid value that was added in r410157.
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5 years agoAST-2014-002: chan_sip: Exit early on bad session timers request
Kinsey Moore [Mon, 10 Mar 2014 13:30:51 +0000 (13:30 +0000)]
AST-2014-002: chan_sip: Exit early on bad session timers request

This change allows chan_sip to avoid creation of the channel and
consumption of associated file descriptors altogether if the inbound
request is going to be rejected anyway.

(closes issue ASTERISK-23373)
Reported by: Corey Farrell
Patches:
     chan_sip-earlier-st-1.8.patch uploaded by Corey Farrell (license 5909)
     chan_sip-earlier-st-11.patch uploaded by Corey Farrell (license 5909)
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Merged revisions 410308 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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5 years agoAST-2014-003: res_pjsip: When handling 401/407 responses don't assume a request will...
Joshua Colp [Mon, 10 Mar 2014 12:53:00 +0000 (12:53 +0000)]
AST-2014-003: res_pjsip: When handling 401/407 responses don't assume a request will have an endpoint.

This change removes the assumption that an outgoing request will always
have an endpoint and makes the authenticate_qualify option work once again.

(closes issue ASTERISK-23210)
Reported by: Joshua Colp
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5 years agopjsip_cli: Create pjsip show channel and contact, and general cli code cleanup.
George Joseph [Sat, 8 Mar 2014 16:50:36 +0000 (16:50 +0000)]
pjsip_cli: Create pjsip show channel and contact, and general cli code cleanup.

Created the 'pjsip show channel' and 'pjsip show contact' commands.
Refactored out the hated ast_hashtab.  Replaced with ao2_container.
Cleaned up function naming.  Internal only, no public name changes.
Cleaned up whitespace and brace formatting in cli code.
Changed some NULL checking from "if"s to ast_asserts.
Fixed some register/unregister ordering to reduce deadlock potential.
Fixed ast_sip_location_add_contact where the 'name' buffer was too short.
Fixed some self-assignment issues in res_pjsip_outbound_registration.

(closes issue ASTERISK-23276)
Review: http://reviewboard.asterisk.org/r/3283/
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5 years agoresource_channels: Check if a passed in ID is NULL before checking its length
Matthew Jordan [Sat, 8 Mar 2014 15:45:59 +0000 (15:45 +0000)]
resource_channels: Check if a passed in ID is NULL before checking its length

Calling strlen on a NULL string is explosive. This patch checks whether or not
the passed in string is NULL or zero length before checking to see if the
string is too long.
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5 years agochan_sip: Fix deadlock of monlock between unload_module and do_monitor
Corey Farrell [Fri, 7 Mar 2014 22:56:15 +0000 (22:56 +0000)]
chan_sip: Fix deadlock of monlock between unload_module and do_monitor

Release monlock before calling pthread_join.  This ensures do_monitor
cannot freeze by locking monlock during module unload.

(closes issue ASTERISK-21406)
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/3284/
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5 years agosorcery: correct field register argument list
Scott Griepentrog [Fri, 7 Mar 2014 22:08:26 +0000 (22:08 +0000)]
sorcery: correct field register argument list

This fixes mistakes I previously made in merging
gtjoseph's changes with mine.
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5 years agoconfig_options: Display the see-also information for CLI config option help
Matthew Jordan [Fri, 7 Mar 2014 21:54:01 +0000 (21:54 +0000)]
config_options: Display the see-also information for CLI config option help

The config option help information has always parsed the <see-also> tags in the
XML documentation. Unfortunately, it just never bothered displaying them on
the CLI. With this patch, when you execute 'config show help [module] [obj]
[option]', it will display what other options are useful to you.

(closes issue ASTERISK-22008)
Reported by: Richard Mudgett
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5 years agores_pjsip: Fix documentation for one touch recording see-also links
Matthew Jordan [Fri, 7 Mar 2014 21:28:12 +0000 (21:28 +0000)]
res_pjsip: Fix documentation for one touch recording see-also links

The one touch recording options have several see-also links between the
various configuration options. These were 'broken' by the snake casing
of those options. This patch corrects the see-also links such that they
reference the correct option names.
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5 years agoMake res_sorcery_realtime filter unknown retrieved results.
Mark Michelson [Fri, 7 Mar 2014 21:23:39 +0000 (21:23 +0000)]
Make res_sorcery_realtime filter unknown retrieved results.

When retrieving data from a database or other realtime backend, it's quite
possible to retrieve variables that Asterisk does not care about but that
are legitimate to exist. Asterisk does not need to throw a hissy fit when
these variables are encountered but rather just filter them out.

Review: https://reviewboard.asterisk.org/r/3305
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5 years agopjsip: allow and disallow show same codecs
Scott Griepentrog [Fri, 7 Mar 2014 21:11:49 +0000 (21:11 +0000)]
pjsip: allow and disallow show same codecs

In order to prevent confusion over the allow and disallow
list of codecs being the same an option for registering a
field as an alias is added.  The alias field will be read
from the configuration file, but afterwards is not listed
as a known field.  With disallow set as an alias, the CLI
command pjsip show endpoint # will list the allow= field,
but not the disallow field.

(closes issue ASTERISK-23092)
Review: https://reviewboard.asterisk.org/r/3193/
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5 years agostasis cache: Enhance to keep track of an item from different entities.
Richard Mudgett [Fri, 7 Mar 2014 20:41:13 +0000 (20:41 +0000)]
stasis cache: Enhance to keep track of an item from different entities.

A stasis cache entry now contains more than a single message/snapshot.  It
contains messages/snapshots for the local entity as well as any remote
entities that post to the cached item.  In addition callbacks can be
supplied when the cache is created to compute and post the aggregate
message/snapshot representing all entities stored in the cache entry.

* All stasis messages now have an eid to indicate what entity posted it.

* The stasis cache enhancements allow device state to cache and aggregate
the device states from local and remote entities in a single operation.
The cached aggregate device state is available immediately after it is
posted to the stasis bus.  This improves performance by eliminating a
cache dump and associated ao2 container traversals to calculate the
aggregate state.

(closes issue ASTERISK-23204)
Reported by: Mark Michelson

Review: https://reviewboard.asterisk.org/r/3281/
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5 years agouniqueid: Fix chan_dahdi, sig_pri, sig_ss7, test_cdr, and test_cel compiler errors.
Richard Mudgett [Fri, 7 Mar 2014 19:19:04 +0000 (19:19 +0000)]
uniqueid: Fix chan_dahdi, sig_pri, sig_ss7, test_cdr, and test_cel compiler errors.

(issue ASTERISK-23120)
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5 years agouniqueid: channel linkedid, ami, ari object creation with id's
Scott Griepentrog [Fri, 7 Mar 2014 15:47:55 +0000 (15:47 +0000)]
uniqueid: channel linkedid, ami, ari object creation with id's

Much needed was a way to assign id to objects on creation, and
much change was necessary to accomplish it.  Channel uniqueids
and linkedids are split into separate string and creation time
components without breaking linkedid propgation.  This allowed
the uniqueid to be specified by the user interface - and those
values are now carried through to channel creation, adding the
assignedids value to every function in the chain including the
channel drivers. For local channels, the second channel can be
specified or left to default to a ;2 suffix of first.  In ARI,
bridge, playback, and snoop objects can also be created with a
specified uniqueid.

Along the way, the args order to allocating channels was fixed
in chan_mgcp and chan_gtalk, and linkedid is no longer lost as
masquerade occurs.

(closes issue ASTERISK-23120)
Review: https://reviewboard.asterisk.org/r/3191/
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5 years agochan_sip: Allow static realtime members to be qualified during module load.
Matthew Jordan [Fri, 7 Mar 2014 05:04:32 +0000 (05:04 +0000)]
chan_sip: Allow static realtime members to be qualified during module load.

When a static realtime peer with qualify=yes is loaded, Asterisk will fail to
send an OPTIONS request due to the lastms being equal to 0. This results in
the peer being unable to receive calls from Asterisk because the status is
permanently UNKNOWN.

This patch allows an OPTIONS request to be sent during module load by
ignoring the lastms value on startup only.

Review: https://reviewboard.asterisk.org/r/3294/

(closes issue ASTERISK-17523)
Reported by: Maciej Krajewski
Tested by: wushumasters
patches:
  realtime_fix_11.7.0.txt uploaded by Trevor Peirce (license 6112)
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5 years agosorcery.c: Fix off-nominal path ref and memory leak in ast_sorcery_objectset_json_cre...
Richard Mudgett [Thu, 6 Mar 2014 23:47:16 +0000 (23:47 +0000)]
sorcery.c: Fix off-nominal path ref and memory leak in ast_sorcery_objectset_json_create().

* Made exit a loop early on error in ast_sorcery_objectset_json_create().

* Removed some dead code in ast_sorcery_objectset_create2().
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5 years agomoh: fix a refcount error with realtime MOH
Russell Bryant [Thu, 6 Mar 2014 23:43:34 +0000 (23:43 +0000)]
moh: fix a refcount error with realtime MOH

I observed a crash in res_musiconhold on an Asterisk 11 system using realtime
MOH.  Investigation of the backtrace showed a corrupt mohclass, implying that
it got destroyed before the code expected it to.  I went looking for reference
counting errors that could have caused this crash and this patch this result.
It contains 2 changes.

1) Remove a usless block of code that was impossible to reach.  There was even
a comment indicating that it was impossible to reach.  The conditional includes
"!ast_test_flag(global_flags, MOH_CACHERTCLASSES)" and it's inside of an if
block with the opposite check "ast_test_flag(global_flags,
MOH_CACHERTCLASSES)".  There's no good reason to keep it around.

2) A similar block to #1 contained a reference counting error.  It stores
state->class in the local variable mohclass without increasing its reference
count.  The reference count on mohclass is decremented at the end of the
function.  This block of code probably very rarely runs, which would help
explain why this system was working fine for many months before experiencing a
crash.

Review: https://reviewboard.asterisk.org/r/3282/
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5 years agosorcery: Create AST_SORCERY dialplan function.
George Joseph [Thu, 6 Mar 2014 22:39:54 +0000 (22:39 +0000)]
sorcery: Create AST_SORCERY dialplan function.

This patch creates the AST_SORCERY dialplan function which allows someone to
retrieve any value from a sorcery-based config file.  It's similar to
AST_CONFIG.

The creation of the function itself was fairly straightforward but it required
changes to the underlying sorcery infrastructure that rippled into individual
sorcery objects.  The changes stemmed from inconsistencies in how sorcery
created ast_variable objectsets from sorcery objects and the inconsistency
in how individual objects used that feature especially when it came to
parameters that can be specified multiple times like contact in aor and match
in identify.  You can read more here...
http://lists.digium.com/pipermail/asterisk-dev/2014-February/065202.html

So, what this patch does, besides actually creating the AST_SORCERY function,
is the following...

* Creates ast_variable_list_append which is a helper to append one ast_variable
  list to another.
* Modifies the ast_sorcery_object_field_register functions to accept the
  already-defined sorcery_fields_handler callback.
* Modifies ast_sorcery_objectset_create to accept a parameter indicating return
  type preference...a single ast_variable with all values concatenated or an
  ast_variable list with multiple entries.  Also fixed a few bugs.
* Modifies individual sorcery object implementations to use the new function
  definition of the ast_sorcery_object_field_register functions.
* Modifies location.c and res_pjsip_endpoint_identifier_ip.c to implement
  sorcery_fields_handler handlers so they return multiple occurrences as an
  ast_variable_list.
* Added a whole bunch of tests to test_sorcery.

(closes issue ASTERISK-22537)
Review: http://reviewboard.asterisk.org/r/3254/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410042 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agopjsip configuration: Make transport TOS values consistent with endpoints
Jonathan Rose [Thu, 6 Mar 2014 19:04:58 +0000 (19:04 +0000)]
pjsip configuration: Make transport TOS values consistent with endpoints

Transport TOS values were interpreted as DSCP values without being documented
as such. Endpoint TOS values (tos_audio/tos_video) behaved normally as TOS
values have historically. This patch makes the transport TOS values behave as
TOS values and makes all TOS values readable as string values (e.g. AF11).
In addition, alembic scripts have been updated to use the proper field types
for all TOS/COS values.

(issue ASTERISK-23235)
Reported by: George Joseph
Review: https://reviewboard.asterisk.org/r/3304/
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5 years agores_stasis_recording: Add a "target_uri" field to recording events.
Joshua Colp [Thu, 6 Mar 2014 18:20:37 +0000 (18:20 +0000)]
res_stasis_recording: Add a "target_uri" field to recording events.

This change adds a target_uri field to the live recording object. It
contains the URI of what is being recorded.

(closes issue ASTERISK-23258)
Reported by: Ben Merrills

Review: https://reviewboard.asterisk.org/r/3299/
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5 years agoDon't attempt to link in an aggregate MWI subscription if an endpoint does not aggreg...
Mark Michelson [Thu, 6 Mar 2014 15:58:13 +0000 (15:58 +0000)]
Don't attempt to link in an aggregate MWI subscription if an endpoint does not aggregate MWI.

Attempting to link a NULL object into an ao2 container had been benign previously, but since
enabling DO_CRASH in the testsuite, this is now causing a crash. It's better to be right
here anyway.
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5 years agoBlocked revisions 410006
George Joseph [Thu, 6 Mar 2014 15:20:51 +0000 (15:20 +0000)]
Blocked revisions 410006

........
sorcery: Create AST_SORCERY dialplan function.

This patch creates the AST_SORCERY dialplan function which allows someone to
retrieve any value from a sorcery-based config file.  It's similar to
AST_CONFIG.

The creation of the function itself was fairly straightforward but it required
changes to the underlying sorcery infrastructure that rippled into individual
sorcery objects.  The changes stemmed from inconsistencies in how sorcery
created ast_variable objectsets from sorcery objects and the inconsistency
in how individual objects used that feature especially when it came to
parameters that can be specified multiple times like contact in aor and match
in identify.  You can read more here...
http://lists.digium.com/pipermail/asterisk-dev/2014-February/065202.html

So, what this patch does, besides actually creating the AST_SORCERY function,
is the following...

* Creates ast_variable_list_append which is a helper to append one ast_variable
  list to another.
* Modifies the ast_sorcery_object_field_register functions to accept the
  already-defined sorcery_fields_handler callback.
* Modifies ast_sorcery_objectset_create to accept a parameter indicating return
  type preference...a single ast_variable with all values concatenated or an
  ast_variable list with multiple entries.  Also fixed a few bugs.
* Modifies individual sorcery object implementations to use the new function
  definition of the ast_sorcery_object_field_register functions.
* Modifies location.c and res_pjsip_endpoint_identifier_ip.c to implement
  sorcery_fields_handler handlers so they return multiple occurrences as an
  ast_variable_list.
* Added a whole bunch of tests to test_sorcery.

(closes issue ASTERISK-22537)
Review: http://reviewboard.asterisk.org/r/3254/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410010 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agores_fax_spandsp: Fix crash when passing ulaw/alaw data to spandsp
Matthew Jordan [Thu, 6 Mar 2014 02:22:59 +0000 (02:22 +0000)]
res_fax_spandsp: Fix crash when passing ulaw/alaw data to spandsp

When acting as a T.38 fax gateway, res_fax_spandsp would at times cause a crash
in libspandsp. This would occur when, during fax tone detection, a ulaw/alaw
frame would be passed to modem_connect_tones_rx. That particular routine
expects the data to be in slin format. This patch looks at the frame type and,
if the data is ulaw/alaw, converts the format to slin before passing it to
modem_connect_tones_rx.

Review: https://reviewboard.asterisk.org/r/3296

(closes issue ASTERISK-20149)
Reported by: Alexandr Gordeev
Tested by: Michal Rybarik
patches:
  spandsp_g711decode.diff uploaded by Michal Rybarik (license 6578)
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5 years agoapp_confbridge: Remove some noop code.
Richard Mudgett [Thu, 6 Mar 2014 00:33:13 +0000 (00:33 +0000)]
app_confbridge: Remove some noop code.
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5 years agores_musiconhold.c: Remove some unnecessary RAII_VAR() usage.
Richard Mudgett [Thu, 6 Mar 2014 00:19:06 +0000 (00:19 +0000)]
res_musiconhold.c: Remove some unnecessary RAII_VAR() usage.

* Made the moh_register() define use useful parameter names.
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5 years agoconfig: Fix inverted test
Kinsey Moore [Wed, 5 Mar 2014 20:41:37 +0000 (20:41 +0000)]
config: Fix inverted test

The test of the result of the stat() call was inverted such that its
output was only used if the call failed. This inverts the test so that
the output of stat() is used correctly. This was causing full reloads
on unchanged files.

(closes issue ASTERISK-23383)
Reported by: David Woolley
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5 years agobridge_native_rtp: Fix crash involving masquerade
Kinsey Moore [Wed, 5 Mar 2014 19:28:31 +0000 (19:28 +0000)]
bridge_native_rtp: Fix crash involving masquerade

It is possible for a channel to be masqueraded out of a bridge which
means it may no longer have RTP glue to check upon leaving said bridge.
If this situation occurred (it's possible at least during dial and call
pickup) then Asterisk would crash. This change makes sure the glue is
checked before use.

(closes issue AST-1290)
Reported by: John Bigelow
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5 years agoalembic: Add missing queue and CDR table creation scripts.
Richard Mudgett [Wed, 5 Mar 2014 18:51:37 +0000 (18:51 +0000)]
alembic: Add missing queue and CDR table creation scripts.

* Added the queues and queue_members tables to the config alembic scripts.

* Added the CDR table alembic creation script.  The CDR table is more of
an example for new setups since the actual table can be fully customized
in cdr_adaptive_odbc.conf.

(closes issue ASTERISK-23233)
Reported by: jmls

Review: https://reviewboard.asterisk.org/r/3227/
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5 years agoFix documentation for PRESENCE_STATE to properly illustrate how to create a presence...
Mark Michelson [Wed, 5 Mar 2014 18:47:30 +0000 (18:47 +0000)]
Fix documentation for PRESENCE_STATE to properly illustrate how to create a presence hint.

There was a missing comma.
This was discovered by Dan Kaplan.
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5 years agoCorrected cross-platform stat nanosecond code
David M. Lee [Wed, 5 Mar 2014 16:58:21 +0000 (16:58 +0000)]
Corrected cross-platform stat nanosecond code

When nanosecond time resolution was added for identifying config file
changes, it didn't cover all of the myriad of ways that one might obtain
nanosecond time resolution off of struct stat.

Rather than complicate the #if even further figuring out one system from
the next, this patch directly tests for the three struct members I know
about today, and #ifdef's accordingly.

Review: https://reviewboard.asterisk.org/r/3273/
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5 years agoFix res/res_http_websocket.c build failure in 32bit due to incorrect print format...
Moises Silva [Wed, 5 Mar 2014 16:26:38 +0000 (16:26 +0000)]
Fix res/res_http_websocket.c build failure in 32bit due to incorrect print format for uint64_t

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409832 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoFix WebRTC over WSS not working
Moises Silva [Wed, 5 Mar 2014 16:22:44 +0000 (16:22 +0000)]
Fix WebRTC over WSS not working

Several fixes for the WebSockets implementation in res/res_http_websocket.c

* Flush the websocket session FILE* as fwrite() may not actually guarantee sending
  the data to the network. If we do not flush, it seems that buffering on the SSL
  socket for outbound messages causes issues

* Refactored ast_websocket_read to take into account that SSL file descriptors
  may be ready to read via fread() but poll() will not actually say so because
  the data was already read from the network buffers and is now in the libc buffers

(closes issue ASTERISK-23099)
(closes issue ASTERISK-21930)
Review: https://reviewboard.asterisk.org/r/3248/
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5 years agoFix references to 'keys' CLI commands in astgenkey
Sean Bright [Wed, 5 Mar 2014 12:06:48 +0000 (12:06 +0000)]
Fix references to 'keys' CLI commands in astgenkey
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5 years agoBlocked revisions 409762
Igor Goncharovskiy [Wed, 5 Mar 2014 06:33:39 +0000 (06:33 +0000)]
Blocked revisions 409762

........
Correct RTP handling in chan_unistim and fix transfer process broken in previous fix:
- Fixed too early RTP setup with phone, that cause no ringback tone on caller side
- Handle call transfer cancel only in STATE_CALL case (related to ASTERISK-23073)

(Reported by: Németh Tamás, niurkin sil)
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5 years agoAdd update_peer function to unistim_rtp_glue, improve other unistim_rtp_glue function...
Igor Goncharovskiy [Wed, 5 Mar 2014 06:17:03 +0000 (06:17 +0000)]
Add update_peer function to unistim_rtp_glue, improve other unistim_rtp_glue functions conforming to other channel drivers. Do not forget auto-detected and user-selected phone settings on 'unistim reload'
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5 years agostasis: Made internal_stasis_subscribe() prototype and definition match exactly.
Richard Mudgett [Wed, 5 Mar 2014 01:05:02 +0000 (01:05 +0000)]
stasis: Made internal_stasis_subscribe() prototype and definition match exactly.
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5 years agofunc_audiohookinheritance: Check If A Channel Was Specified
Michael L. Young [Tue, 4 Mar 2014 19:34:52 +0000 (19:34 +0000)]
func_audiohookinheritance: Check If A Channel Was Specified

This patch prevents a crash when using the function audiohookinheritance without
setting the channel.

(closes issue ASTERISK-23104)
Reported by: Joel Vandal
Tested by: Joel Vandal
Patches:
    asterisk-23104_audiohook_inherit_no_channel-11.diff
                                     uploaded by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/3272/
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5 years agores_rtp_asterisk: Fix one way audio problems with hold/unhold when using ICE
Jonathan Rose [Tue, 4 Mar 2014 17:22:32 +0000 (17:22 +0000)]
res_rtp_asterisk: Fix one way audio problems with hold/unhold when using ICE

ICE sessions will now be restarted if sessions are changed to use new sets of
remote candidates.

(closes issue ASTERISK-22911)
Reported by: Vytis Valentinavičius
Review: https://reviewboard.asterisk.org/r/3275/
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5 years agoAO2: Add an assert for bad objects
Kinsey Moore [Tue, 4 Mar 2014 16:55:43 +0000 (16:55 +0000)]
AO2: Add an assert for bad objects

This adds an assert that will only be active if Asterisk is compiled
with DO_CRASH and allows the testsuite to fail tests that would
otherwise require log file parsing.
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5 years agoMinor whitespace change to 'sip show peers' output.
Sean Bright [Tue, 4 Mar 2014 14:55:24 +0000 (14:55 +0000)]
Minor whitespace change to 'sip show peers' output.

(closes issue ASTERISK-23406)
Reported by: ibercom
Tested by: ibercom
Patches:
    asterisk-11.patch uploaded by ibercom
........

Merged revisions 409472 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 409473 from http://svn.asterisk.org/svn/asterisk/branches/11
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5 years agores_stasis_recording: Fix memory leak of the absolute name.
Joshua Colp [Mon, 3 Mar 2014 19:44:58 +0000 (19:44 +0000)]
res_stasis_recording: Fix memory leak of the absolute name.
........

Merged revisions 409422 from http://svn.asterisk.org/svn/asterisk/branches/12

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5 years agodoxygen: Tweak the link back to ye olde Digium website
Matthew Jordan [Mon, 3 Mar 2014 02:08:58 +0000 (02:08 +0000)]
doxygen: Tweak the link back to ye olde Digium website
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5 years agoMakefile: replace -O6 with -O3
Tzafrir Cohen [Sun, 2 Mar 2014 17:03:56 +0000 (17:03 +0000)]
Makefile: replace -O6 with -O3

-O6 is not a legal option of gcc. Unofficially gcc considers it to be
equivalent of -O3. clang chalks on it, though. This commit sets the
default optimization flag to be -O3, like gcc actually considered it.

Review: https://reviewboard.asterisk.org/r/3280/
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5 years agores_pjsip_session: Set options (100rel, timers) on incoming sessions.
Joshua Colp [Sat, 1 Mar 2014 20:28:04 +0000 (20:28 +0000)]
res_pjsip_session: Set options (100rel, timers) on incoming sessions.

This change passes options to the UAS creation function. This in turn
sets up 100rel and session timer properties on the incoming session.

Reported by Julian Russell on asterisk-users mailing list.
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Merged revisions 409287 from http://svn.asterisk.org/svn/asterisk/branches/12

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5 years agodevicestate.c: Simplified some logic in _ast_device_state().
Richard Mudgett [Sat, 1 Mar 2014 00:05:38 +0000 (00:05 +0000)]
devicestate.c: Simplified some logic in _ast_device_state().
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5 years agostasis_cache.c: Remove some unnecessary RAII_VAR() usage.
Richard Mudgett [Sat, 1 Mar 2014 00:02:02 +0000 (00:02 +0000)]
stasis_cache.c: Remove some unnecessary RAII_VAR() usage.
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5 years agostasis.c: Misc code cleanups.
Richard Mudgett [Fri, 28 Feb 2014 23:31:58 +0000 (23:31 +0000)]
stasis.c: Misc code cleanups.

* Remove some unnecessary RAII_VAR() usage.

* Made the struct stasis_subscription ao2 object use the ao2 lock instead
of a redundant join_lock in the struct for ast_cond_wait().

* Removed locks on some ao2 objects that don't need the lock.

* Made the topic pool entries container use the ao2 template functions.

* Add some missing allocation failure checks.

* Add missing cleanup in off nominal path of dispatch_message().
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409271 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agochan_sip: Add precautionary p->owner checks.
Richard Mudgett [Fri, 28 Feb 2014 21:44:26 +0000 (21:44 +0000)]
chan_sip: Add precautionary p->owner checks.

* Add precautionary p->owner checks in sip_hangup(), get_refer_info(),
get_also_info(), and interpret_t38_parameters().

* Simplify some tangled logic in get_refer_info(), get_also_info(), and
add_rpid().

* Removed some dead code in handle_request_invite().

(closes issue ASTERISK-23323)
Reported by: Walter Doekes
Patches:
      issueA23323-more_p_owner_checks-1.8.x.patch (license #5674) uploaded by wdoekes (modified)
      issueA23323-more_p_owner_checks-11.x.patch (license #5674) uploaded by wdoekes (modified)
      issueA23323-more_p_owner_checks-12.x.patch (license #5674) uploaded by wdoekes (modified)
      issueA23323-more_p_owner_checks-trunk.patch (license #5674) uploaded by wdoekes (modified)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409257 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoapp_queue: Fix documented AMI event name
Kinsey Moore [Fri, 28 Feb 2014 21:24:47 +0000 (21:24 +0000)]
app_queue: Fix documented AMI event name

During the rewrite of AMI events to use the Stasis bus, the name of the
QueueMemberPaused event was changed to QueueMemberPause. This corrects
documentation to reflect that.
........

Merged revisions 409234 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409237 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agochan_sip: Fix crash in ast_channel_hangupcause_set().
Richard Mudgett [Fri, 28 Feb 2014 18:03:56 +0000 (18:03 +0000)]
chan_sip: Fix crash in ast_channel_hangupcause_set().

* Fix crash in ast_channel_hangupcause_set() because p->owner not checked
before calling.  Regression introduced by the fix for ASTERISK-22621.

(closes issue ASTERISK-23135)
Reported by: OK

(issue ASTERISK-23323)
Reported by: Walter Doekes
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Merged revisions 409156 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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5 years agoMultiple revisions 409129-409130
Jonathan Rose [Thu, 27 Feb 2014 19:54:19 +0000 (19:54 +0000)]
Multiple revisions 409129-409130

........
  r409129 | jrose | 2014-02-27 13:19:02 -0600 (Thu, 27 Feb 2014) | 15 lines

  res_rtp_asterisk: Fix checklist creating problems in ICE sessions

  Prior to this patch, local candidate lists including SRFLX would fail to start
  properly when building ICE candidate check lists. This patch fixes that problem
  by making sure that each SRFLX candidate is associated with the proper
  base address so that the check list can create matches properly.
  This patch was written by jcolp. The issue will be left open to await testing
  by the issue participants.

  (issue ASTERISK-23213)
  Reported by: Andrea Suisani
  Review: https://reviewboard.asterisk.org/r/3256/
........
  r409130 | jrose | 2014-02-27 13:38:10 -0600 (Thu, 27 Feb 2014) | 8 lines

  res_rtp_asterisk: correct build error from r409129

  Accidentally placed a declaration below functional code

  (issue ASTERISK-23213)
  Reported by: Andrea Suisani
  Review: https://reviewboard.asterisk.org/r/3256/
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Merged revisions 409129-409130 from http://svn.asterisk.org/svn/asterisk/branches/11
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409132 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoFix memory stomping bug in astman.
David M. Lee [Thu, 27 Feb 2014 16:26:18 +0000 (16:26 +0000)]
Fix memory stomping bug in astman.

This memset complained in dev mod on my Ubuntu box. The memset is both
unnecessary and dangerous. At this point, m hasn't been initialized
yet, so the memset will write off to whatever address happens to be
on the stack at the time.
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Merged revisions 409077 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 409083 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 409087 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409091 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agores_fax: Comment out default settings from res_fax.conf.
Corey Farrell [Thu, 27 Feb 2014 16:08:03 +0000 (16:08 +0000)]
res_fax: Comment out default settings from res_fax.conf.

Comment out many settings in res_fax.conf.sample. The defaults are set in
res_fax.c, so setting the same value in sample config does nothing but make
the sample config more fragile.

(closes issue ASTERISK-23231)
Reported by: David Brillert
Review: https://reviewboard.asterisk.org/r/3261/
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Merged revisions 409053 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 409054 from http://svn.asterisk.org/svn/asterisk/branches/12

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5 years agores_pjsip_sdp_rtp: Apply packetization rules on inbound SDP handling
Matthew Jordan [Thu, 27 Feb 2014 12:29:56 +0000 (12:29 +0000)]
res_pjsip_sdp_rtp: Apply packetization rules on inbound SDP handling

The setting 'use_ptime' is supposed to tell Asterisk to honour the ptime
attribute in an offer, preferring it to whatever packetization
preferences have been set internally. Currently, however, something
rather quirky will happen:

(1) The SDP answer will be constructed in create_outgoing_sdp_stream.
    This will use the preferences from the endpoint, such that the 200 OK
    response will add the packetization preferences from the endpoint, and
    not what was offered.
(2) When the 200 response is issued, apply_negotiated_sdp_stream is called.
    This will call apply_packetization, which will use the ptime attribute
    from the offer internally.

We end up telling the offerer to use the internal ptime attribute, but we end
up using the offered ptime attribute. Hilarity ensues.

This patch modifies the behaviour by calling apply_packetization from
negotiate_incoming_sdp_stream, which is called prior to
create_outgoing_sdp_stream. This causes the format preferences on the
session's media object to be set to the inbound ptime value (if 'use_ptime'
is enabled), such that the construction of the answer gets the right value
immediately.

Review: https://reviewboard.asterisk.org/r/3244/
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Merged revisions 408999 from http://svn.asterisk.org/svn/asterisk/branches/12

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5 years agotest_stasis.c: Misc cleanups.
Richard Mudgett [Wed, 26 Feb 2014 23:35:20 +0000 (23:35 +0000)]
test_stasis.c: Misc cleanups.

* Make the consumer ao2 object use the ao2 lock instead of a redundant
lock in the struct for ast_cond_wait().

* Fixed some curly brace placements.

* Fixed use of malloc(0).  malloc(0) has variant behavior.  It is up to
the implementation to determine if it returns NULL or a valid pointer that
can be later passed to free().
........

Merged revisions 408983 from http://svn.asterisk.org/svn/asterisk/branches/12

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5 years agopjsip: avoid edge case potential crash in answer()
Scott Griepentrog [Wed, 26 Feb 2014 19:00:56 +0000 (19:00 +0000)]
pjsip: avoid edge case potential crash in answer()

When accidentally compiling against a wrong version of
pjsip headers with a different pjsip_inv_session size,
the invite_tsx structure could be null in the answer()
function.  This led to a crash because it attempted to
send the session response with an uninitialized packet
pointer.  This patch presets packet to null and adds a
diagnostic log message to explain why the call fails.

Review: https://reviewboard.asterisk.org/r/3267/
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Merged revisions 408970 from http://svn.asterisk.org/svn/asterisk/branches/12

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5 years agores_ari: Make some additional error responses consistent with the rest of the system.
Joshua Colp [Wed, 26 Feb 2014 17:04:38 +0000 (17:04 +0000)]
res_ari: Make some additional error responses consistent with the rest of the system.

This change makes some error cases use ast_ari_response_error to construct their
error responses instead of manually doing it. This ensures they are consistent
with the other error responses.

Based on the original patch as done by Paul Belanger on the associated review.

Review: https://reviewboard.asterisk.org/r/2904/
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5 years agoPJSIP: Fix some bad spacing
Kinsey Moore [Wed, 26 Feb 2014 13:47:49 +0000 (13:47 +0000)]
PJSIP: Fix some bad spacing
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5 years agoPJSIP: Prevent crash if channel has gone away
Kinsey Moore [Wed, 26 Feb 2014 13:45:12 +0000 (13:45 +0000)]
PJSIP: Prevent crash if channel has gone away

It is currently possible for an ast_sip_session to exist without an
associated channel as is the case when a new invite is coming in or
just after a hangup is issued on a chan_pjsip channel. Part of the
attended transfer code assumed the channel would be non-NULL and used
it as such causing a crash. This bug was exposed thanks to the attended
transfer ARI test in the test suite.

(closes issue ASTERISK-23287)
Reported by: Matt Jordan
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Merged revisions 408941 from http://svn.asterisk.org/svn/asterisk/branches/12

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5 years agoImplement functions handling keypress, display icons and text for i2004 KEM support.
Igor Goncharovskiy [Wed, 26 Feb 2014 08:57:14 +0000 (08:57 +0000)]
Implement functions handling keypress, display icons and text for i2004 KEM support.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408932 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agores_pjsip_exten_state: Presence for digium phones
Kevin Harwell [Tue, 25 Feb 2014 17:51:51 +0000 (17:51 +0000)]
res_pjsip_exten_state: Presence for digium phones

Added presence support for digium phones.

Review: https://reviewboard.asterisk.org/r/3239/
........

Merged revisions 408882 from http://svn.asterisk.org/svn/asterisk/branches/12

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5 years agores_pjsip_send_to_voicemail: transferring to voicemail for digium phones
Kevin Harwell [Tue, 25 Feb 2014 17:47:06 +0000 (17:47 +0000)]
res_pjsip_send_to_voicemail: transferring to voicemail for digium phones

Added the ability for transferring directly to voicemail on digium phones.
Added a new module that checks for the presence of a custom header and/or
diversion header within a sip REFER.  If either is found and they specify
a sending to voicemail action then variables are added to the channel
allowing the user access to them in the dialplan.  Dialplan can then be
written that branches based upon these values allowing, for instace, for
a single number to be used for dialing and/or accessing voicemail directly.

Also fixed a problem where the PJSIP_HEADER function was allowing non pjsip
channels through (checked to make sure it has the correct channel type before
proceeding).

Review: https://reviewboard.asterisk.org/r/3245/
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Merged revisions 408880 from http://svn.asterisk.org/svn/asterisk/branches/12

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5 years agoconfigs/voicemail.conf.sample - Make mailcmd sample text more explicit
Rusty Newton [Tue, 25 Feb 2014 17:44:53 +0000 (17:44 +0000)]
configs/voicemail.conf.sample - Make mailcmd sample text more explicit

Made the wording a bit more explicit. Didn't really change the meaning.
........

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Merged revisions 408878 from http://svn.asterisk.org/svn/asterisk/branches/12

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5 years agomain: Initialize dialplan providing core components prior to module pre-load
Matthew Jordan [Sat, 22 Feb 2014 23:31:10 +0000 (23:31 +0000)]
main: Initialize dialplan providing core components prior to module pre-load

It is possible to pre-load pbx_config. As a result, pbx_config - which will
load and parse the dialplan - will attempt to use various dialplan components,
such as device state providers and presence state providers, prior to them
being initialized by the core. This would lead to a crash, as the components
had not created their Stasis cache entries.

This patch moves a number of core component initializations before the module
pre-load. This guarantees that if someone does pre-load pbx_config - or other
pbx modules - that the Stasis caches for the various core components are
created.

(closes issue ASTERISK-23320)
Reported by: xrobau

(closes issue ASTERISK-23265)
Reported by: Andrew Nagy
Tested by: Andrew Nagy, Rusty Newton
........

Merged revisions 408855 from http://svn.asterisk.org/svn/asterisk/branches/12

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5 years agoignore AST_CONTROL_PVT_CAUSE_CODE without any messages
Alexandr Anikin [Sat, 22 Feb 2014 18:01:20 +0000 (18:01 +0000)]
ignore AST_CONTROL_PVT_CAUSE_CODE without any messages

(closes issue ASTERISK-23336)
Reported by: Alexander Semych
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Merged revisions 408839 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408840 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoRemove extra defines of AST_PBX_MAX_STACK.
Corey Farrell [Sat, 22 Feb 2014 02:31:04 +0000 (02:31 +0000)]
Remove extra defines of AST_PBX_MAX_STACK.

* Ensure AST_PBX_MAX_STACK is only defined in extconf.h and pbx.h.
* Fix incorrect function parameters in utils/extconf.c.

(closes issue ASTERISK-23141)
Reported by: Maxim
Review: https://reviewboard.asterisk.org/r/3241/
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Merged revisions 408785 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 408787 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408788 65c4cc65-6c06-0410-ace0-fbb531ad65f3