ICE/STUN/TURN support in res_rtp_asterisk is also now optional.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382900 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In ASTERISK-17888, the AMI Registry event during SIP registrations was supposed
to include the Username field. Somehow, one of the events was missed. This
patch corrects that - the Username field should be included in all AMI Registry
events involving SIP registrations.

(issue ASTERISK-17888)

(closes issue ASTERISK-21201)
Reported by: Dmitriy Serov
patches:
  chan_sip.c.diff uploaded by Dmitriy Serov (license 6479)
........

Merged revisions 382847 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 382848 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382852 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fix issue with 'unistim show info' CLI command when device connected not configured
........

Merged revisions 382827 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382828 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Added an option "discard_remote_hold_retrieval" (default "no") that if set does
not trigger the music on hold event.  This essentially stops telling the peer
to start music on hold.

(issue ABE-2899)
Reported by: Denis Alberto Martinez
Review: https://reviewboard.asterisk.org/r/2336/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382787 65c4cc65-6c06-0410-ace0-fbb531ad65f3

struct conference_bridge_user -> struct confbridge_user
struct conference_bridge -> struct confbridge_conference
struct conference_state -> struct confbridge_state

struct conference_bridge_user *conference_bridge_user -> struct confbridge_user *user
struct conference_bridge_user *cbu -> struct confbridge_user *user
struct conference_bridge *conference_bridge -> struct confbridge_conference *conference

The names are now generally shorter, consistently used, and don't conflict
with the struct names.

This patch handles the renaming part of the issue.

(issue ASTERISK-20776)
Reported by: rmudgett


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382764 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Prior to this change, certain conditions for sending the message would
result in an address of '(null)' being used in the via header of the
SIP message because a NULl value of pvt->ourip was used when initially
generating the via header. This is fixed by adding a call to build_via
when the address is set before sending the message.

(closes issue ASTERISK-21148)
Reported by: Zhi Cheng
Patches:
	700-sip_msg_send_via_fix.patch uploaded by Zhi Cheng (license 6475)
........

Merged revisions 382739 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382746 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(issue ASTERISK-20887)
(issue ASTERISK-20959)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382724 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fixes test failure introduced in r382685.

(issue ASTERISK-20887)
(issue ASTERISK-20959)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382721 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The cache dump mechanism allows the developer to retreive multiple
items of a given type (or of all types) from the cache residing in a
stasis caching topic in addition to the existing single-item cache
retreival mechanism.  This also adds to the caching unit tests to
ensure that the new cache dump mechanism is functioning properly.

Review: https://reviewboard.asterisk.org/r/2367/
(issue ASTERISK-21097)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382705 65c4cc65-6c06-0410-ace0-fbb531ad65f3

For the initial use of this bus, I took some work kmoore did creating
channel snapshots. So rather than create AMI events directly in the
channel code, this patch generates Stasis events, which manager.c uses
to then publish the AMI event.

This message bus provides a generic publish/subscribe mechanism within
Asterisk. This message bus is:

 - Loosely coupled; new message types can be added in seperate modules.
 - Easy to use; publishing and subscribing are straightforward
   operations.

In addition to basic publish/subscribe, the patch also provides
mechanisms for message forwarding, and for message caching.

(issue ASTERISK-20887)
(closes issue ASTERISK-20959)
Review: https://reviewboard.asterisk.org/r/2339/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382685 65c4cc65-6c06-0410-ace0-fbb531ad65f3

After r382670, get_ip_and_port_from_sdp was no longer used.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382671 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Originally, way back in r201583, we added the alternate RTP address so
that the RTP engine would expect to receive audio from a new source
when a glare re-INVITE occurred. In r382589, we remove the alternate
RTP source, as the 'secret' probation mode allows for switching to a new
RTP source when a previous source stops sending RTP. At the time, it
seemed appropriate to set the RTP source based on the information in the
glared re-INVITE.

Unfortunately, that doesn't work so well - in a glared re-INVITE that occurs
with no SDP - such as in a connected line update that glances - we'll set
the RTP source to an invalid address. In subsequent re-INVITE requests from
this Asterisk instance, we'll then send an invalid media address, which will
result in the remote side sending a 488. Whoops.

There isn't any need to reset the RTP source - if we're using strictrtp, we'll
simply synchronize to a new source when we stop getting packets from the old
one. If we aren't using strictrtp, then again there shouldn't be a problem.

Note that the Asterisk Test Suite's connectedline test caught this error.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382670 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382648 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382636 65c4cc65-6c06-0410-ace0-fbb531ad65f3

r381835 fixed a bug in vm_mailbox_snapshot where combining INBOX and Old forgot
that Urgent also "counts" as new messages. This fixed the problem when any of
the three folders was specified and the combine option was used.

It missed the case where the folder isn't specified and we build a snapshot of
all folders. This patch corrects that.
........

Merged revisions 382617 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382621 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Another instance of attribute retrieval not being freed properly.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382604 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Many places that allocated to pull out an attribute are now freed
properly.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382600 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Often, Asterisk may realize that a change in the source of an RTP stream is
about to occur and ask that the RTP engine reset it's lock on the current RTP
source. In certain scenarios, it may take awhile for the new remote system to
send RTP packets, while the old remote system may continue providing RTP during
that time period. This causes Asterisk to re-lock onto the old source, thereby
rejecting the new source when the old source stops sending RTP and the new
source begins.

This patch prevents that by having a constant secondary, 'secret' probation
mode enabled when an RTP source has been chosen. RTP packets from other sources
are always considered, but never chosen unless the current RTP source stops
sending RTP.

Review: https://reviewboard.asterisk.org/r/2364

(closes issue AST-1124)
Reported by: John Bigelow
Tested by: John Bigelow

(closes issue AST-1125)
Reported by: John Bigelow
Tested by: John Bigelow
........

Merged revisions 382573 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382589 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Strings retrieved via ast_xml_get_text() must be freed with
ast_xml_free_text().


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382587 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Messages sent while the logger thread is shutting down will now have
their associated callid freed properly.
........

Merged revisions 382574 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382575 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If ast_threadpool_set_size with a size equal to the current size, a
reference to a set_size_data structure would be leaked.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382555 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Ownership of the listener reference is not transferred because the
listener is reffed when placed into the taskprocessor. Ensure that the
listener is dereffed properly.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382489 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds support for RFC 3327 "Path" headers. This can be enabled in
sip.conf using the 'supportpath' setting, either on a global basis or on a
peer basis. This setting enables Asterisk to route outgoing out-of-dialog
requests via a set of proxies by using a pre-loaded route-set defined by the
Path headers in the REGISTER request. This patch also adds Realtime support
for dynamically updating the Path information for a peer.

A huge thank-you to Klaus Darillion and Olle E Johansson for their efforts
in writing this patch.

Review: https://reviewboard.asterisk.org/r/2235/
Review: https://reviewboard.asterisk.org/r/991/

(closes issue ASTERISK-16884)
Reported by: klaus3000
Tested by: klaus3000, oej, mjordan
patches:
  path-1.8.0-patch.txt uploaded by klaus3000 (License 5054)
  oolong-path-support-trunk in team branch by oej (License 5267)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382440 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fix several unreleased mutex locks that cause problem with processing calls
Reported by: Daniel Bohling
Tested by: Daniel Bohling

(Closes issue ASTERISK-21119)
........

Merged revisions 382409 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 382410 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382411 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382392 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Several new IEs were not given types (or names), causing the comparison
function to improperly succeed.  This adds those.

(closes issue AST-1128)
........

Merged revisions 382390 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382391 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This fix checks to make sure that if a confbridge record start command is issued
from the CLI it will always use the file name given on the CLI even if it
changes between start/stop records for a conference.  Previously it had been
reusing the same file between start/stops even if a new filename was given.

(issue AST-1088)
Reported by: John Bigelow
........

Merged revisions 382385 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382386 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/2332/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382340 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The original report had to do with a realtime peer behind NAT being pruned and
the peer's private address being used instead of its external address.  Upon
debugging, it was discovered that this was being caused by the addition of
the auto_force_rport and auto_comedia settings.

This patch does the following:

* Adds a missing note to the CHANGES file indicating that the default global nat
  setting is auto_force_rport

* Constify the 'req' parameter for check_via()

* Add calls to check_via() in a couple of places in order for the auto_*
  settings to do their job in attempting to determine if NAT is involved

* Set the flags SIP_NAT_FORCE_RPORT and SIP_PAGE2_SYMMETRICRTP if the auto_*
  settings are in use where it was needed

* Moves the copying of peer flags up in build_peer() to before they are used;
  this fixes the realtime prune issue

* Update the contrib/realtime schemas to allow the nat column to handle the
  different nat setting combinations we have

This patch received a review and "Ship It!" on the issue itself.

(closes issue ASTERISK-20904)
Reported by: JoshE
Tested by: JoshE, Michael L. Young
Patches:
  asterisk-20904-nat-auto-and-rt-peersv2.diff Michael L. Young (license 5026)
........

Merged revisions 382322 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382323 65c4cc65-6c06-0410-ace0-fbb531ad65f3

While the ICE negotiation is occurring leave strictrtp in an open state, media can and will come from different places.
........

Merged revisions 382298 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382299 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If the end result of the ICE negotiation resulted in the path for media
changing it was possible for the strictrtp code to discard the RTP packets.
This change causes strictrtp to enter learning mode once again when the
ICE negotiation has completed successfully.
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Merged revisions 382296 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382297 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382295 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382294 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This was causing build failures against external libraries that happened to use
them, unless silly hacks were added to the modules that used those headers.

Review: https://reviewboard.asterisk.org/r/2359/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382292 65c4cc65-6c06-0410-ace0-fbb531ad65f3

A deadlock can occur in chan_iax2 when it attempts to set the caller ID, as it
already holds the iax2 private lock and improperly fails to obtain the channel
lock before calling ast_set_callerid. By not safely obtaining the channel lock,
a locking inversion can take place, causing a deadlock.

This patch solves this by calling the required deadlock avoidance functions
that obtain the channel lock before setting the caller ID.

Thanks to Pavel for fixing my syntax errors and testing this patch out.

(closes issue ASTERISK-21128)
Reported by: Pavel Troller
Tested by: Pavel Troller
patches:
  ASTERISK-21128-1.8.diff uploaded by mjordan (license 6283)
  ASTERISK-21128-modified-1.8.diff uploaded by Pavel Troller (license 6302)
........

Merged revisions 382233 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 382234 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382236 65c4cc65-6c06-0410-ace0-fbb531ad65f3

For some channel drivers, specifically those that have a varying rate in the
number of audio samples, the audio quality for a MeetMe conference can be
exceedingly poor. This is due to a unilateral application of the DENOISE
function in func_speex to channels joining the conference.

The denoiser function in the speex library is initialized with the number of
audio samples in each sample that will be provided to it. If the number of
audio samples changes, the denoiser has to be thrown away and re-initialized.

While this could be worked around by removing func_speex, that doesn't help
if you actually use the denoiser with other channels on the system.

This patches does the following:
 * Checks for the presence of func_speex as opposed to codec_speex when
   determining if the DENOISE function is present (which is where the function
   is actually implemented)
 * Adds an option to MeetMe 'n' that causes the denoiser to not be applied
   to a channel when it joins. This keeps the current behavior the default, but
   let's users disable the denoiser if it causes problems on their system.

Review: https://reviewboard.asterisk.org/r/2358

(closes issue AST-1062)
Reported by: Thomas Arimont
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Merged revisions 382227 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 382230 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382232 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382204 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382203 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-20638)
Reported by: eelcob
Patches:
      pedantic-call-pickup-from-tag.patch uploaded by eelcob (license 6442)
........

Merged revisions 382171 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 382174 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382182 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* The powerpcspe Linux port uses linux-gnuspe as the OS string.
* Our build system shouldn't really care for that, so just call it linux-gnu.
* Original report: Roland Stigge , http://bugs.debian.org/701505

Review: https://reviewboard.asterisk.org/r/2357/
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Merged revisions 382110 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 382111 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382113 65c4cc65-6c06-0410-ace0-fbb531ad65f3