Commit Graph

25510 Commits

Author SHA1 Message Date
Richard Mudgett c8ebf3e3c7 Revert -r411073. It didn't help and blew up the system.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411087 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-25 15:47:17 +00:00
Richard Mudgett 89e12de79d locking: Add temporary sanity checks.
Add some temporary sanity checks to hunt for locking problems with the
masquerade supertest.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411073 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-24 23:36:36 +00:00
Joshua Colp 6d81951f0d chan_sip: Always use fromdomain if set for domain, even if callerid is set to restricted.
(closes issue ASTERISK-20841)
Reported by: Kelly Goedert
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Merged revisions 411021 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 411022 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 411023 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411024 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-24 21:39:46 +00:00
Richard Mudgett 236d17362d res_pjsip_registrar.c: Miscellaneous cleanup in rx_task().
* Fix variable shadowing of 'updated' by renaming it to 'contact_update'.

* Checked 'contact_update' for ast_sorcery_copy() failure.

* Removed silly use of RAII_VAR() for 'contact_update'.
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Merged revisions 410995 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410996 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-21 16:04:09 +00:00
Sean Bright b44d324891 Make the AEL load process less chatty.
Switched a bunch of LOG_NOTICEs to ast_debug.  This time without breaking the
build.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410994 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-21 15:50:11 +00:00
Sean Bright 14942ecb17 Revert r410981. aelparse blew up.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410993 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-21 15:30:37 +00:00
Sean Bright df2d959d7d Remove a LOG_NOTICE from ast_config_engine_register.
There is enough indication from the CLI that we are loading a realtime engine
as it is.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410982 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-21 15:16:50 +00:00
Sean Bright 922d0b7565 Make the AEL load process less chatty.
Switched a bunch of LOG_NOTICEs to ast_debug.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410981 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-21 15:14:13 +00:00
Jonathan Rose 3c16865fc2 app_confbridge: Fix bug - users with startmuted set don't start muted
(closes issue ASTERISK-23461)
Reported by: Chico Manobela
Review: https://reviewboard.asterisk.org/r/3373/
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Merged revisions 410965 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 410966 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410967 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-20 23:02:45 +00:00
Richard Mudgett 1ba13718fc assigned-uniqueids: Miscellaneous cleanup and fixes.
* Fix memory leak in ast_unreal_new_channels().  Made it generate the ;2
uniqueid on a stack variable instead of mallocing it.

* Made send error response to ARI and AMI requests instead of just logging
excessive uniqueid length and allowing truncation.  action_originate() and
ari_channels_handle_originate_with_id().

* Fixed minor truncating uniqueid hole when generating the ;2 uniqueid
string length.  Created public and internal lengths of uniqueid.  The
internal length can handle a max public uniqueid plus an appended ;2.

* free() and ast_free() are NULL tolerant so they don't need a NULL test
before calling.

* Made use better struct initialization format instead of the position
dependent initialization format.  Also anything not explicitly initialized
in the struct is initialized to zero by the compiler.

* Made ast_channel_internal_set_fake_ids() use the safer
ast_copy_string() instead of strncpy().

Review: https://reviewboard.asterisk.org/r/3371/
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Merged revisions 410949 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410950 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-20 16:35:57 +00:00
Mark Michelson 57239bfe37 PJSIP: Allow for identify sections to be specified in sorcery.conf.
"identify" is a special type of configuration object in PJSIP because
unlike the other objects, it is not provided by the base res_pjsip module.
Instead, it is provided by the res_pjsip_endpoint_identifier_ip module. If
using the default sorcery wizard (config,criteria=type=identify) then things
work because the module that applies the default wizard is the correct module.

However, if attempting to use sorcery.conf to apply an alternate wizard, it
was not possible. If you attempted to specify the identify object type in the
res_pjsip section, then the object could not be registered since the object
was undocumented for the res_pjsip module. There was no alternate configuration
section defined for it, so you were out of luck if you wanted to override the
default wizard.

With this change, the identify section will properly have a sorcery.conf-based
wizard applied when the identify definition is within the res_pjsip_endpoint_identifier_ip
section.
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Merged revisions 410933 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410934 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-19 17:27:57 +00:00
Joshua Colp 326153d949 res_stasis: Fix a bug where the default bridge type was not set.
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Merged revisions 410918 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410919 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-19 14:25:31 +00:00
Joshua Colp 1cf74b8776 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


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2014-03-19 12:54:25 +00:00
Matthew Jordan e33e003f78 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
2014-03-19 02:33:55 +00:00
Rusty Newton 35fb3a564b 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
2014-03-18 23:32:00 +00:00
Scott Griepentrog eecb74a9a7 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
2014-03-18 15:45:04 +00:00
Matthew Jordan 77db94a25a 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
2014-03-18 15:28:45 +00:00
Joshua Colp 216b04e6f4 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
2014-03-18 12:45:49 +00:00
Sean Bright 8357027080 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
2014-03-18 11:52:15 +00:00
Richard Mudgett 614b6abc38 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
2014-03-18 02:09:25 +00:00
Joshua Colp cc40bf5317 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
2014-03-17 22:54:32 +00:00
Joshua Colp 932fb5a6e2 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/
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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
2014-03-17 22:46:56 +00:00
Russ Meyerriecks ed50ef4dc8 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)

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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
2014-03-17 22:24:03 +00:00
Mark Michelson eba91d2a98 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.
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Merged revisions 410696 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410699 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-17 19:35:17 +00:00
Mark Michelson d44aefeef4 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
2014-03-17 17:22:12 +00:00
Richard Mudgett 1900bae7b6 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
2014-03-17 16:48:55 +00:00
Matthew Jordan 2c5484c869 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


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410651 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-16 20:27:28 +00:00
Russell Bryant a0a51a65e0 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
2014-03-15 15:24:23 +00:00
Mark Michelson 478c7faf0b Fix failing realtime sorcery tests.
The store realtime callback needs to return a positive value for
sorcery to treat the store as a success.
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Merged revisions 410625 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410626 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-14 21:56:21 +00:00
Jonathan Rose 3a565767d7 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


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410624 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-14 21:36:55 +00:00
Mark Michelson 510a6e6e64 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


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2014-03-14 20:55:06 +00:00
Mark Michelson 9665c2d3eb 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


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2014-03-14 18:11:55 +00:00
Mark Michelson 1fc33bc795 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


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2014-03-14 18:05:04 +00:00
Scott Griepentrog 9e9707f90f 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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2014-03-14 17:56:53 +00:00
Jonathan Rose ff63012c4e 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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2014-03-14 16:42:54 +00:00
Mark Michelson c1e9d2f177 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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2014-03-14 16:19:21 +00:00
Jonathan Rose 4c2b1c225b 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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2014-03-14 16:17:26 +00:00
Richard Mudgett 66718a06f7 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/
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Merged revisions 410555 from http://svn.asterisk.org/svn/asterisk/branches/12


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2014-03-14 16:01:13 +00:00
Richard Mudgett 251868dc57 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


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2014-03-13 21:27:15 +00:00
Kinsey Moore 5247a0d990 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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2014-03-13 19:33:22 +00:00
Joshua Colp 1b5c098976 Multiple revisions 410509-410510
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  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.
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  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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2014-03-13 13:25:09 +00:00
Richard Mudgett f627a0aca0 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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2014-03-12 19:06:52 +00:00
Richard Mudgett de3dc17cc5 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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2014-03-12 18:47:10 +00:00
Joshua Colp d00c1ac23e 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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2014-03-12 12:51:34 +00:00
Joshua Colp 2fa1ff6e75 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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2014-03-11 16:07:42 +00:00
Richard Mudgett 7c854d65af 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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2014-03-10 17:21:01 +00:00
Scott Griepentrog ef69b5176d 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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2014-03-10 16:33:10 +00:00
Kinsey Moore c300f7e5a8 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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2014-03-10 13:30:51 +00:00
Joshua Colp aa57dcf634 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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2014-03-10 12:53:00 +00:00
George Joseph 3ff60b75b1 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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2014-03-08 16:50:36 +00:00