Commit Graph

23080 Commits

Author SHA1 Message Date
Jonathan Rose 68c63d7965 manager: remove curses dependent stuff from r375103
Upon further examination, this code was causing compliation problems on
CentOS at the least (possibly on any machine without curses) and also
the local value of COLS is used even with a remote console, so it is
less than ideal.

(issue ASTERISK-20396)
Reported by: Johan Wilfer


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375175 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-17 20:34:40 +00:00
Pedro Kiefer 8b34dc8192 Adds new formats to app_alarmreceiver, ALAW calls support and enhanced protection.
Commiting this on behalf of Kaloyan Kovachev (license 5506).
AlarmReceiver now supports the following DTMF signaling types:
 - ContactId
 - 4x1
 - 4x2
 - High Speed
 - Super Fast
We are also auto-detecting which signaling is being received. So support for
those protocols should work out-the-box. Correctly identify ALAW / ULAW calls.
Some enhanced protection for broken panels and malicious callers where added.

(closes issue ASTERISK-20289)
Reported by: Kaloyan Kovachev

Review: https://reviewboard.asterisk.org/r/2088/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375150 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-17 19:02:46 +00:00
Kinsey Moore 372e29620c Ensure Asterisk fails TCP/TLS SIP calls when certificate checking fails
When placing a call to a TCP/TLS SIP endpoint whose certificate is not
signed by a configured CA certificate, Asterisk would issue a warning
and continue to process the call as if there was not an issue with the
certificate.  Asterisk now properly fails the call if the certificate
fails verification or if the certificate does not exist when
certificate checking is enabled (the default behavior).

(closes issue ASTERISK-20559)
Reported by: kmoore

Review: https://reviewboard.asterisk.org/r/2163/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375149 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-17 19:01:27 +00:00
Walter Doekes 6d57ecd48c Change a few warnings to debug and the inverse.
Remove the "RTP Read too short" warning for RTP keepalives. Remove the
the warning about the application delimiter switch from pipe to comma.
(You should've done this by now.) Make cdr_odbc report more when an
insert fails. Make chan_sip warn less when the peer wants SRTP (and we
don't) or sends a zero port to disable a media type.

Review: https://reviewboard.asterisk.org/r/2167
(closes issue ASTERISK-20538)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375137 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-17 14:24:52 +00:00
Walter Doekes 1a0646aec1 Fixes to the fd-oriented SIP TCP reads.
Don't crash on large user input. Allow SIP headers without space.
Optimize code a bit.

Review: https://reviewboard.asterisk.org/r/2162
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375114 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-16 21:46:09 +00:00
Walter Doekes 8a65f47e88 Don't do SIP contact/route DNS if we're not using the result.
In many cases (for peers behind NAT or for TCP sockets) we do not need
to look up any hostname in the Contact (or Route) when sending an
in-dialog request. This should reduce netsock2.c: getaddrinfo errors in
certain scenarios.

Review: https://reviewboard.asterisk.org/r/2156


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375110 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-16 21:38:00 +00:00
Jonathan Rose b2f9542f61 manager: Change display of 'manager show commands' and 'manager show command'
manager show commands now shows the full name of the command being displayed
regardless of size. The privilege column has also been removed from this
display. It will also now use the full length of the terminal if curses is
available. Manager show command will now always display the privilege of
the manager command within the CLI.

(closes ASTERISK-20396)
Reported by: Johan Wilfer
Review: https://reviewboard.asterisk.org/r/2143/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375103 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-16 20:45:49 +00:00
Pedro Kiefer ef60d0efc0 Fixes two small regressions from ASTERISK-20157
- receive_dtmf_digits had the wrong buffer length
- app_alarmreceiver should wait 100ms before sending the second part of handshake

(closes issue ASTERISK-20484)
Reported by: Jean-Philippe Lord
Tested by: Jean-Philippe Lord, Pedro Kiefer
Patches:
     ASTERISK-20484_v2.diff uploaded by Kaloyan Kovachev (license 5506)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375081 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-16 19:26:20 +00:00
Walter Doekes 2142fc3bc7 Update sip_request_call SIP dial string documentation.
This was missed when merging review r1859.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375080 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-16 19:25:11 +00:00
Joshua Colp c4df9778cb Remove a log message that was left in accidentally from call-id logging development.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-16 14:09:39 +00:00
Mark Michelson e9ab568f88 Fix some potential misuses of ast_str in the code.
Passing an ast_str pointer by value that then calls
ast_str_set(), ast_str_set_va(), ast_str_append(), or
ast_str_append_va() can result in the pointer originally
passed by value being invalidated if the ast_str had
to be reallocated.

This fixes places in the code that do this. Only the
example in ccss.c could result in pointer invalidation
though since the other cases use a stack-allocated ast_str
and cannot be reallocated.

I've also updated the doxygen in strings.h to include
notes about potential misuse of the functions mentioned
previously.

Review: https://reviewboard.asterisk.org/r/2161
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-15 21:25:29 +00:00
Igor Goncharovskiy e41a591dfc Fix underscreen buttons warnings apeared while transfer process
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375017 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-15 08:26:58 +00:00
Andrew Latham d6dd24d71e Doxygen Updates
Update and extend the configuration_file group and enable linking.

(issue ASTERISK-20259)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375009 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-14 21:59:22 +00:00
Andrew Latham 00a9699e05 Doxygen Updates
Update and extend the configuration_file group and enable linking.

(issue ASTERISK-20259)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375008 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-14 21:57:42 +00:00
Andrew Latham b106b77041 Title update
Update title that was left behind many years ago. Used revision 6596 as my guide for what it should be.

(issue ASTERISK-20259)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375007 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-14 21:56:13 +00:00
Andrew Latham 3820f1586e Doxygen Updates - Title update
Update and extend the configuration_file group and enable linking.  Update title that was left behind many years ago.

(issue ASTERISK-20259)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375006 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-14 21:47:40 +00:00
Andrew Latham 02bcd5bfb6 Doxygen Updates - Title update
Update and extend the configuration_file group and enable linking.  Update title that was left behind many years ago.

(issue ASTERISK-20259)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375005 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-14 21:46:45 +00:00
Andrew Latham cfc6f60ca3 Doxygen Updates - Title update
Update and extend the configuration_file group and enable linking to the application.  Update title that was left behind many years ago.

(issue ASTERISK-20259)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375004 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-14 21:45:16 +00:00
Andrew Latham c7857504df Doxygen Updates - Title update
Update and extend the configuration_file group and enable linking to the resource.  Update title that was left behind many years ago.

(issue ASTERISK-20259)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375003 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-14 21:44:27 +00:00
Tzafrir Cohen 707423e6aa Update config.guess and config.sub: 2012-10-10
Update config.guess and config.sub to revision
fb456b34ef4aa02b95dc6be69aaa66fa94a844fb from the savannah.gnu.org git
repo. Adds support for e.g. aarch64 (ARM 64bit).

config.guess:timestamp='2012-09-25'
config.sub:timestamp='2012-10-10'
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374996 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-14 12:23:46 +00:00
Andrew Latham 4ac2c1148b Update CREDITS
Update Jean-Denis and add myself

(issue ASTERISK-20259)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374970 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-13 19:58:20 +00:00
Andrew Latham ec0747d4f6 Multiplatform Makefile Update
Paul Belanger pointed out that using sed in the Makefile is an issue with multiple platforms.  We are cleaning up the Doxygen config as a following step so I just switched the sed inplace changes to be an echo append instead.

(issue ASTERISK-20259)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374963 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-13 17:38:41 +00:00
Andrew Latham e51432027a Doxygen Clean ups
Add app_skel.c as an example in app.c and fix some formating for the "Dial Privacy scripts" so it actually shows up in the Doxygen output.

(issue ASTERISK-20259)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374956 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-13 16:38:48 +00:00
Andrew Latham 4c2711d4db Test for Asterisk Version info
Doxygen uses the ASTERISKVERSION as a sub header. If a SVN export is done and no .svn or .version file exists it defualts to UNKNOWN__and_probably_unsupported which is honest but not great for the online docs.  During the "make progdocs" I added a test for this and just warned and ommitted the version.

(issue ASTERISK-20259)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374949 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-13 16:05:12 +00:00
Andrew Latham 4040875cfa Correct output directory
During testing I used an alternate output directory and mistakenly committed it.  Matt Jordan noticed and I reverted.  This is the correct setting for local output to match with all branches.

(issue ASTERISK-20259)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374948 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-13 15:40:40 +00:00
Andrew Latham 447d2e55dd Add licens/copyright header
Begin update of static-http files and general clean ups.  This only adds the standard header to the files.

(issue ASTERISK-20503)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374941 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-13 15:14:51 +00:00
Andrew Latham 83b13ebc04 Add check for Doxygen
The autoconf configuration system had a test for DOT but not for Doxygen.  I added the test for Doxygen and did an overhaul of the Makefile check to a much simpler process.

(issue ASTERISK-20259)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374940 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-13 15:06:39 +00:00
Kinsey Moore 0eab8b669d Avoid a segfault on invalid format names
If a format name was not found by ast_getformatbyname, a NULL pointer
would be passed into ast_format_rate and immediately dereferenced.
This ensures that a valid pointer is used since the structure is
already allocated on the stack.

(closes issue DPH-523)
Reported-by: Steve Pitts
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374933 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-12 21:58:29 +00:00
Mark Michelson c7b23cbb0a Do not use a FILE handle when doing SIP TCP reads.
This is used to solve an issue where a poll on a file
descriptor does not necessarily correspond to the readiness
of a FILE handle to be read.

This change makes it so that for TCP connections, we do a
recv() on the file descriptor instead.

Because TCP does not guarantee that an entire message or even
just one single message will arrive during a read, a loop has
been introduced to ensure that we only attempt to handle a
single message at a time. The tcptls_session_instance structure
has also had an overflow buffer added to it so that if more
than one TCP message arrives in one go, there is a place to
throw the excess.

Huge thanks goes out to Walter Doekes for doing extensive review
on this change and finding edge cases where code could fail.

(closes issue ASTERISK-20212)
reported by Phil Ciccone

Review: https://reviewboard.asterisk.org/r/2123
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374924 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-12 16:31:01 +00:00
Andrew Latham 399428224d Append Doxygen to Debian packages list
Add Doxygen to the Debian install list.  I will check for other platforms like Red Hat

(issue ASTERISK-20259)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374897 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-11 23:40:44 +00:00
Andrew Latham e3246f1e8f Update JQuery URL to recent version
The JQuery URL to version 1.4 will be removed within the life span of Asterisk 11.  This is a compatible upgrade by using the URL for 1.8.

(issue ASTERISK-20503)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374889 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-11 22:43:52 +00:00
Andrew Latham 7226606f77 Continue to group config files
(issue ASTERISK-20259)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374888 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-11 22:39:02 +00:00
Andrew Latham 42a8af5c9a CREDITS clean up
As discussed online http://lists.digium.com/pipermail/asterisk-dev/2012-October/057245.html the credits file needs some cleaning.  This is 95% whitespace with a few additions found in file headers.  Further additions should be added here instead of in the file being updated.

(issue ASTERISK-20259)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-11 22:35:41 +00:00
Andrew Latham 5627cd6f6b Revert Local testing Config
Revert a local testing config that I made.  This was not intended to be committed.
Thank you Matt Jordan for noticing this.

(issue ASTERISK-20259)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374879 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-11 21:40:02 +00:00
Joshua Colp ccb7b3a1b5 Fix a bug where audio on Google Voice would not work due to ignoring candidates.
Instead of ignoring parts of the message that are not known just ignore the ones
we know may be present and that would cause a problem.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374878 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-11 21:19:33 +00:00
Joshua Colp cd9745be1b Fix an issue where outgoing calls would fail to establish audio due to ICE negotiation failures.
This change removes the requirement for ufrag and pwd in the transport stanza and also
makes us the controlling agent.

(closes issue ASTERISK-20554)
Reported by: mmichelson

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374852 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-11 16:06:28 +00:00
Mark Michelson 825607e09b Don't make chan_sip export global symbols.
During testing, it was discovered that having chan_sip
export global symbols was problematic.

The biggest problem was that load order was affected.
Trying to use realtime could be problematic since in
all likelihood the necessary realtime driver(s) would
not be loaded before chan_sip.

In addition, it was found that it was impossible to
use the Digium Phone Module for Asterisk since it
must be loaded before chan_sip since it must hook
into chan_sip's configuration parsing.

The solution is to use a virtual table in the same
manner that other modules in Asterisk do, like
app_voicemail.

(closes issue ASTERISK-20545)
Reported by: kmoore

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374849 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-11 15:49:02 +00:00
Matthew Jordan 5ac43a08ad Fix incorrect billing duration reported when batch mode is enabled
Similar to r369351, the billing duration can be skewed when batch mode is
enabled.  This happened much more rarely than the duration, as it only
occured when the call was answered (thereby indicating an actual answer
time) and immediately hung up on (indicating a billsec of 0).  Since
a billing time of '0' can either mean that the call immediately ended
or that the CDR was improperly answered, we have to use additional information
to know whether or not we can trust the CDR billsec value.  Prior to this
patch, we looked to see if we had a valid answer time.  If we did, and
billsec was zero, we used the current time to calculate what billsec value
we could from the CDR being written.  If batch mode is enabled, this will
incorrectly report a billsec value being much greater than the actual
duration of the call.

Instead of relying on the presence of an answer time to know whether or not
we can re-calculate the billsec for the CDR, we now also use the presence
of the CDR's end time to know if we need to re-calculate or whether we can
trust the billsec value that we have.  This prevents erroneous jumps in the
billsec value, while still making sure that in the worst case, some billing
time will be calculated.

(closes issue AST-1016)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
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2012-10-11 15:44:38 +00:00
Joshua Colp 755c2b8708 Consider the Google Talk content stanza name (jin:content) valid.
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2012-10-11 13:34:52 +00:00
Richard Mudgett 516c9ec665 app_queue: Made pass connected line updates from the caller to ringing queue members.
Party A calls Party B
Party B puts Party A on hold.
Party B calls a queue.
Ringing queue member D sees Party B identification.
Party B transfers Party A to the queue.
Queue member D does not get a connected line update for Party A.
Queue member D answers the call and still sees Party B information.

However, if Party A later transfers the call to Party C then queue member
D gets a connected line update for Party C.

* Made pass connected line updates from the caller to queue members while
the queue members are ringing.

(closes issue AST-1017)
Reported by: Thomas Arimont

(closes issue ABE-2886)
Reported by: Thomas Arimont
Tested by: rmudgett

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2012-10-10 21:05:51 +00:00
Kinsey Moore 357b733b66 Fix segfault regression from r370681
Due to usage of ast_hook_send_action, AMI action handling code should
be able to handle a NULL mansession->session.  This would cause a crash
on NULL dereference if action_originate was called from
ast_hook_send_action.

(closes issue ASTERISK-20544)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374793 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-10 13:40:40 +00:00
Richard Mudgett 9a0ff62452 Fix execution of 'i' extension due to uninitialized variable.
The fix for ASTERISK-18243 added code that could potentially use
dst_exten[] uninitialized.  As a result the 'i' exten may not be executed
when it should.

(closes issue ASTERISK-20455)
Reported by: Richard Miller
Patches:
      pbx-1.8.16.0.diff (license #5685) patch uploaded by Richard Miller
      Made some cosmetic modifications.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374778 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-09 22:24:10 +00:00
Joshua Colp 766d133c62 Improve logging for DTLS-SRTP failure situations.
(closes issue ASTERISK-20487)
Reported by: mjordan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374757 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-09 21:35:53 +00:00
Richard Mudgett 3e1d2917bb dahdi.conf.sample: Add description for "buffers" setting.
This contains an edited version of the patch originally created by John
Bigelow.

(closes issue ASTERISK-14435)
Reported by: John Bigelow
Patches:
      buffers.patch (license #5091) patch uploaded by John Bigelow
      0001-dahdi.conf.sample-Add-description-for-buffers-settin.patch (license #5417) patch uploaded by Shaun Ruffell
      Modified
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374730 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-08 22:31:09 +00:00
Richard Mudgett ab548bb0ce Fix deletion of unopenable spool files.
If scan_service() cannot open the spool file, it logs a message saying
that it will delete the file and calls remove_from_queue() to do it.
However, remove_from_queue() fails to delete the spool file because struct
outgoing has not yet been fully initialized.

* Merged allocating a new struct outgoing and init_outgoing() into
new_outgoing().  Allocation is initialization.

* Made apply_outgoing() not initialize the spool filename in struct
outgoing.

* Made apply_outgoing() call ast_trim_blanks() and ast_skip_blanks()
rather than manually inlining them.

* Reduced indentation levels in apply_outgoing().

* Fixed a garbled comment in remove_from_queue().

* Reworked scan_service() to simplify it.

(closes issue ASTERISK-17231)
Reported by: David Chappell
Patches:
      spool_open_failure.diff (license #4997) patch uploaded by David Chappell
      Started with this patch.
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* Fixed some memory leaks on off nominal paths in init_outgoing() when
merging into the new_outgoing() function dealing with o->capabilities.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374717 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-08 21:24:11 +00:00
Matthew Jordan 3620fcff36 Disable ICE support by default
Since there are a number of legacy devices out there that fail to handle ICE
candidates properly (which is a nice way of saying something much uglier),
disable it by default.

Support for ICE candidates can be enabled in rtp.conf using the icesupport
setting.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374677 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-08 20:39:26 +00:00
Matthew Jordan be906d6318 Resolve issues in ConfBridge regarding marked, waitmarked, and unmarked users
Thank's to Neil Tallim (flan)'s tireless testing, issue reporting, and patches
it became clear that app_confbridge had some complex logic in how it handled
interactions between marked, waitmarked, and unmarked users.  In particular,
there were some areas in which the interactions between the users resulted
in inconsistent behavior, and app_confbridge was missing logic in how to handle
some corner cases.  Some areas included:
 * Poor handling of mixing unmarked and waitmarked users
 * Inconsistencies in how MOH and muting was applied to various users
 * Handling of various announcements for different user profile options
flan's patches seem to fix the various issues, but highlighted how hard the
code could be to maintain.  In an attempt to make things easier to maintain and
to more fully enumerate the various cases that exist, this patch breaks up the
logic into a state machine-like setup.

Please note that the various state transitioned are documented on the Asterisk
wiki:

https://wiki.asterisk.org/wiki/display/AST/Confbridge+state+changes

Review: //https://reviewboard.asterisk.org/r/2072/

Note that for the following issues, mjordan uploaded the patch, although it
was written by twilson.  Any contributor license discrepency is due to that.

(closes issue ASTERISK-19562)
Reported by: flan
Tested by: flan, mjordan, jrose
patches:
  bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by twilson (license 6283)

(closes issue ASTERISK-19726)
Reported by: flan
Tested by: flan
patches:
  bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by twilson (license 6283)

(closes issue ASTERISK-20181)
Reported by: Jonathan White
Tested by: Jonathan White
patches:
  bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by twilson (license 6283)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374658 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-08 18:48:34 +00:00
Matthew Jordan 35b12af8b6 pjproject: Fix for Solaris builds. Do not undef s_addr.
pjproject, in order to solve build problems on Windows [1], undefines s_addr in
one of it's headers that is included in res_rtp_asterisk.c. On Solaris s_addr
is not a structure member, but defined to map to the real strucuture member,
therefore when building on Solaris it's possible to get build errors like:

    [CC] res_rtp_asterisk.c -> res_rtp_asterisk.o
    In file included from /export/home/admin/asterisk-11-svn/include/asterisk/stun.h:29,
                     from res_rtp_asterisk.c:51:
    /export/home/admin/asterisk-11-svn/include/asterisk/network.h: In function `inaddrcmp':
    /export/home/admin/asterisk-11-svn/include/asterisk/network.h:92: error: structure has no member named `s_addr'
    /export/home/admin/asterisk-11-svn/include/asterisk/network.h:92: error: structure has no member named `s_addr'
    res_rtp_asterisk.c: In function `ast_rtp_on_ice_tx_pkt':
    res_rtp_asterisk.c:706: warning: dereferencing type-punned pointer will break strict-aliasing rules
    res_rtp_asterisk.c:710: warning: dereferencing type-punned pointer will break strict-aliasing rules
    res_rtp_asterisk.c: In function `rtp_add_candidates_to_ice':
    res_rtp_asterisk.c:1085: error: structure has no member named `s_addr'
    make[2]: *** [res_rtp_asterisk.o] Error 1
    make[1]: *** [res] Error 2
    make[1]: Leaving directory `/export/home/admin/asterisk-11-svn'
    gmake: *** [_cleantest_all] Error 2

Unfortunately, in order to make this work, I also had to make sure pjproject
only used the typdef pj_in_addr and not the struct pj_in_addr so that when
building Asterisk I could "typedef struct in_addr pj_in_addr". It's possible
then that the library and users of those interfaces in Asterisk have a different
idea about the type of the argument, while on the surface it looks like they are
all 32 bit big endian values.

[1] http://trac.pjsip.org/repos/changeset/484

(issues ASTERISK-20366)
Reported by: Ben Klang
Tested by: Ben Klang, mjordan
patches:
  0001-pjproject-Fix-for-Solaris-builds.-Do-not-undef-s.patch uploaded by Shaun Ruffell (license 5417)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374643 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-08 00:45:36 +00:00
Matthew Jordan 6b5677a41d Trivial patch to make 'best_score' defined for all architectures.
Fixes trivial build error on Solaris:

  acl.c: In function `get_local_address':
  acl.c:196: error: `best_score' undeclared (first use in this function)
  acl.c:196: error: (Each undeclared identifier is reported only once
  acl.c:196: error: for each function it appears in.)
  make[2]: *** [acl.o] Error 1

(issue ASTERISK-20366)
Reported by: Ben Klang
Tested by: Ben Klang
patches:
  0002-main-acl.c-Trivial.-best_score-should-be-defined-for.patch by Shaun Ruffell (license 5417)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374633 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-07 17:33:38 +00:00
Matthew Jordan bd36827e98 Handle capability stanzas that fail to provide node or version information
While XEP-0115 states that the node and ver attributes are both required, some
devices fail to provide either field.  Prior to this patch, failure to provide
the node or ver attribute would cause a crash in res_xmpp.  While failing to
provide the node or ver attribute is technically invalid, since this
information is not utilized by Asterisk except for reporting purposes, for
interoperability reasons, we continue to process the capability stanza anyways.

(closes issue ASTERISK-20495)
Reported by: Martin W
Tested by: Martin W
patches:
  20495.patch uploaded by Martin W (license #6434)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-06 03:22:37 +00:00