Add packet 0x014A (variable length call info messages) to skinny for newer
firmware. Plenty of unknown information but includes the equivalent functionality
as the fixed size callinfo packet already included.
Only send this packet if protocol reported is >= 17.
Review: https://reviewboard.asterisk.org/r/2290/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380755 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Previously, Asterisk only processed session timer information if both the
'Supported: timer' and 'Session-Expires' headers were present. However, the
Session-Expires header is optional. If we were to receive a request with a
Min-SE greater than our configured session-expires, we would respond with a
'Session-Expires' header that was too small.
This patch cleans the situation up a bit, always processing timer information
if the 'Supported: timer' header is present.
(closes issue ASTERISK-20787)
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/2299/
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This patch is mostly a reorganization of existing code with a few exceptions:
* Added doxygen comments to all of the extracted functions.
* Split reload_firmware(int unload) into iax_firmware_reload() and
iax_firmware_unload() for readability.
* Create iax_firmware_traverse() to support the 'iax2 show firmware' CLI
command.
* Renamed iax_check_version() to iax_firmware_get_version() and change its
arguments and return value so that it returns a success/failure value and sets
the selected version into an out parameter to avoid confusion with failure and
version 0.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380695 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r380671 | qwell | 2013-01-31 12:59:28 -0600 (Thu, 31 Jan 2013) | 4 lines
Remove a cross-compile workaround.
ar and ranlib can be easily detected with autoconf.
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r380672 | qwell | 2013-01-31 13:00:38 -0600 (Thu, 31 Jan 2013) | 2 lines
Always check for libm, regardless of configure options.
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r380673 | qwell | 2013-01-31 13:03:03 -0600 (Thu, 31 Jan 2013) | 7 lines
Add support for parallel builds of pjproject.
Also adds proper dependency checking, and direct .a file targets. We don't
take advantage of this currently, but we will soon.
(issue ASTERISK-20815)
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* Fixed the potential of losing the multiplexed bridge thread when the
last channel leaves and another joins while the multiplexed thread is
being shut down.
* Refactored and improved the management of the serviced channels array.
* Changed the channels count to a bridges count so it only needs to be
incremented rather than changed by two.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380666 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Building Asterisk on Raspbian with hard-float support fails as it uses the
string 'linux-gnueabihf' for host os, as opposed to 'linux-gnueabi'. This patch
modifies the configure script for Asterisk such that it will match on any
string beginning with 'linux-gnueabi', as opposed to requiring an explicit
match.
(closes issue ASTERISK-21006)
Reported by: Christian Hesse
Tested by: Christian Hesse
patches:
linux-gnueabihf.patch uploaded by Christian Hesse (license 6459)
linux-gnueabihf-autoconf.patch uploaded by Christian Hesse (license 6459)
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A user in #asterisk ran into a problem where a configuration error prevented
the chan_sip module from being loaded. Upon fixing their configuratione error,
they could no longer load the chan_sip module. This was because the
configuration checking happened after the SIP provider was registered with the
Asterisk core, and subsequent attempts to load the SIP module failed as the
provider was already registered.
Since we want to detect any failure in registering chan_sip as early as
possible (as that could be emblematic of a deeper mismatch between module
and Asterisk core), this patch does not change the registration location, but
does ensure that if a module load is declined, we unregister the module as
the SIP api provider.
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RFC5347 section 2.5.2 states the following:
...
The attribute "T38MaxBitRate" was once incorrectly registered with
IANA as "T38maxBitRate" (lower-case "m"). In accordance with T.38
examples and common implementation practice, the form "T38MaxBitRate"
SHOULD be generated by implementations conforming to this package.
In general, it is RECOMMENDED that implementations of this package
accept lowercase, uppercase, and mixed upper/lowercase encodings of
all the T.38 attributes.
...
Asterisk currently does not perform case insensitive matching on the T.38
attributes. This causes the T38MaxBitRate attribute to be negotiated at
2400 baud instead of 14400 (or whatever value you actually wanted).
This patch makes it so that when we compare T.38 attributes, we do so in a case
insensitive fashion.
Note that while the issue reporter did not directly write the patch, they
contributed to it (and would have provided one themselves if the license had
gone through a tad faster), and hence get attribution for it.
Review: https://reviewboard.asterisk.org/r/2298/
(closes issue ASTERISK-20897)
Reported by: Eric Hill
Tested by: Eric Hill
patches:
-- uploaded by Eric Hill
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The ICalendar module had a systemic memory leak on each fetch of data from
the ICalendar source. The previous fetched data was not being properly
disposed. This patch makes it so that before each fetch of data, we dispose
of the previously fetched data.
(closes issue ASTERISK-21012)
Reported by: Joel Vandal
Tested by: Joel Vandal
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This patch just moves the IAX2 source and header files into a separate iax2
sub-directory in the channels directory, similar to how the sip source files are
structured.
The only thing that was added was an #ifndef to protect provision.h from multiple
inclusion.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380433 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Multiple channels logging in as the same agent can result in dead channels
waiting for a condition signal that will never come because another
channel thread stole it. A symptom is chan_sip repeatedly generating
warning messages about rescheduling autodestruction of dialogs with an
agent channel owner.
* Made only login_exec() (the app AgentLogin) clear the agent_pvt->chan
pointer to prevent multiple channels from logging in as the same agent.
agent_read(), agent_call(), and agent_set_base_channel() no longer
disconnect the agent channel from the agent_pvt. This also eliminates the
need to keep checking for agent_pvt->chan being NULL.
* Made agent_hangup() not wake up the AgentLogin agent thread until it is
done.
* Made agent_request() not able to get the agent until he has logged in
and any wrapup time has expired.
* Made agent_request() use ast_hangup() instead of agent_hangup() to
correctly dispose of a channel.
* Removed agent_set_base_channel(). Nobody calls it and it is a bad thing
in general.
* Made only agent_devicestate() determine the current device state of an
agent. Note: Agent group device states have never been supported.
Review: https://reviewboard.asterisk.org/r/2260/
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The original fix (r380043) for getting Asterisk to respond with the correct
tag overlooked some corner cases, and the fact that the same code is in 1.8.
This patch moves the building of the crypto line out of
sdp_crypto_process(). Instead, it merely copies the accepted tag. The call to
sdp_crypto_offer() will build the crypto line in all cases now, using a tag of
"1" in the case of sending offers.
(closes issue ASTERISK-20849)
Reported by: José Luis Millán
Review: https://reviewboard.asterisk.org/r/2295/
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A regression was introduced which removed automatic fallback behavior from
the PBX. This behavior was used by call parking (or at least documented as
how the feature works) in order to select an extension when the flat channel
extension wasn't available from the comebackcontext. Parking now handles
the fallbacks internally in order to keep behavior matching with how it is
documented.
(closes issue ASTERISK-20716)
Reported by: Chris Gentle
Review: https://reviewboard.asterisk.org/r/2296/
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In r369028, chan_sip's processing of media streams in an SDP was modified to
better handle multiple offered media streams. Part of that change modified
how streams were declined. Previously, declined media streams were not
handled in an RFC compliant manner; now, we set the port number to 0 in the
media stream definition and proceed on with the next media stream.
Unfortunately, the formatting of the declined media stream forgot to append a
'\r\n' to the end of the media stream. This is normally added to the accepted
media streams later on in the processing of the SDP. Since the declined media
stream uses a different buffer than the accepted media streams (and is a
malloc'd buffer as opposed to a struct ast_str), it's easier to just slap the
'\r\n' on the declined media stream buffer rather than attempt to append it
later on.
So, that's what we do. And now some devices (and probably some providers) will
be a bit happier (but probably not terribly happy, since we just rejected
something they offered).
Review: https://reviewboard.asterisk.org/r/2297/
(closes issue ASTERISK-20908)
Reported by: Dennis DeDonatis
Tested by: Dennis DeDonatis
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With ptlib 2.10.9, the configure script fails due to grep returning multiple
matches for the pattern it searches for. This patch updates the pattern
matching to return only the actual version for the symbol searched for,
PTLIB_VERSION.
(closes issue ASTERISK-20980)
Reported by: Stefan Reuter
patches:
ASTERISK-20980-1.patch uploaded by Stefan Reuter (license 5339)
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This patch came about due to a problem observed where wav files had an
empty header. The header is supposed to be updated in wav_close(). It
turns out that this was broken when the cache_record_files option from
asterisk.conf was enabled. The cleanup code was moving the file to its
final destination *before* running the close() method of the file
destructor, so the header didn't get updated.
Another problem here is that the move was being done before actually
closing the FILE *.
Finally, the last bug fixed here is that I noticed that wav_close()
checks for stream->filename to be non-NULL. In the previous cleanup
order, it's checking a pointer to freed memory. This doesn't actually
cause anything to break, but it's treading on dangerous waters. Now the
free() of stream->filename is happening after the format module's
close() method gets called, so it's safer.
Review: https://reviewboard.asterisk.org/r/2286/
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The "sound_only_one" sound was not being set even though it was configured. In
looking into this, I found that the "join" and "leave" prompts were not being
set either.
(closes issue ASTERISK-20898)
Reported by: Stephan
Tested by: Stephan
Patches:
asterisk-20898-custom-sounds-ignored.diff uploaded by
Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2289/
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* Rename multiplexed_thread variables to muxed_thread. It is shorter and
my editer tagging works much better. Struct names and variable names have
different purposes and therefore should have different names.
* Renamed the multiplexed_threads container to muxed_threads for
consistency.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380142 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Made multiplexed_bridge_destroy() check if anything to destroy and
cleared bridge_pvt pointer after destruction.
* Made multiplexed_add_or_remove() handling of the chans array simpler.
* Extracted bridge_channel_poke().
* Simplified bridge_array_remove() handling of the bridge->array[]. The
array does not have a NULL sentinel pointer.
* Made ast_bridge_new() not create a temporary bridge just to see if it
can be done. Only need to check if there is an appropriate bridge tech
available.
* Made ast_bridge_new() clean up on allocation failures.
* Made destroy_bridge() free resources in the opposite order of creation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380109 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Sorcery is a unifying data access layer which provides a pluggable mechanism to allow
object creation, retrieval, updating, and deletion using different backends (or wizards).
This is a fancy way of saying "one interface to rule them all" where them is configuration,
realtime, and anything else that comes along.
Review: https://reviewboard.asterisk.org/r/2259/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380069 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Adds a dial softkey when the device is in DAFD. The softkey is greyed (unusable)
until a possible dialplan match is entered. Code includes updating
transmit_selectsoftkeys to allow the use of a button mask. Also add option
to use # or * as a dial now button. Original patch by snuffy cleaned up by myself.
Review: https://reviewboard.asterisk.org/r/2277/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380057 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When Asterisk responds with an SDP ANSWER for SRTP, it had the code to
correctly fill in the crypto data, which was overwritten by a call to
sdp_crypto_offer. Corrected the situation by changing sdp_crypto_offer
to not replacing crypto data if it already exists.
(closes issue ASTERISK-20849)
Reported by: José Luis Millán
Tested by: Iñaki Baz Castillo
Patches:
fix_sdp_crypto_tags.diff uploaded by Pedro Kiefer (license 6407)
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The documentation for ConfbridgeList states that the Conference field is
optional. That's not really the case: if you fail to provide a Conference
number, the command will kick back an error.
(closes issue AST-1090)
Reported by: John Bigelow
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This adds the ability to get the DPMA version, a listing of the local
firmware directory, and indexes of configured remote directories.
(closes issue AST-1070)
Reported By: Malcolm Davenport
Tested By: Kinsey Moore <kmoore@digium.com>
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380004 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes the problem, but the issue includes a test which is still
being considered for the automated test suite.
(issue ASTERISK-20919)
Reported by: NITESH BANSAL
Patches:
patch_ast_fax_spandsp.patch uploaded by NITESH BANSAL (license 6418)
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During the conversion to the newer CLI command structure the old definitions were
commented out. I think it's safe to remove them completely now.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379936 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The old prompts for the administrator menu were inadequate. They didn't mention
that the menu had additional options through the 8 key and pressing the 8 key
wouldn't reveal what those options were. This patch fixes all of that while
also organizing code pertaining to each individual menu type which was
previously all stored in one gigantic function along with many of the basic
conference functions.
(closes issue AST-996)
Reported by: John Bigelow
Review: http://reviewboard.digium.internal/r/360/
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This patch adds the capability for asynchronous manipulation of audio being
played back to a channel though a new AMI action "ControlPlayback". The
ControlPlayback action supports a number of operations, the availability of
which depend on the application being used to send audio to the channel.
When the audio playback was initiated using the ControlPlayback application
or CONTROL STREAM FILE AGI command, the audio can be paused, stopped,
restarted, reversed, or skipped forward. When initiated by other mechanisms
(such as the Playback application), the audio can be stopped, reversed, or
skipped forward.
Review: https://reviewboard.asterisk.org/r/2265/
(closes issue ASTERISK-20882)
Reported by: mjordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes two bugs:
* If an outbound call is made from a SLA phone using SLAStation, then there is
no ringtone audible to the phone that originates the call. The indication of
the ringing was not being passed to the SLA station; this patch fixes that
by passing through the progress indications.
* If an SLA station hangs up before the called party answers, then the channel
to the called party continues to ring until a timeout occurs. If the called
party manages to answer, Asterisk attempts to connect the called party to
a non-existant MeetMe room. This patch corrects the behavior by abandoning
the call attempt if it detects that the SLA station is no longer in use
while attempting to call the called party.
Review: https://reviewboard.asterisk.org/r/2275/
(closes issue ASTERISK-20462)
Reported by: dkerr
patches:
asterisk-11-bugid20440+20462.patch uploaded by dkerr (license 5558)
asterisk-11-bugid20462.patch uploaded by dkerr (license 5558)
(closes issue ASTERISK-20440)
Reported by: dkerr
patches:
asterisk-11-bugid20440.patch uploaded by dkerr (license 5558)
asterisk-11-bugid20440+20462.patch uploaded by dkerr (license 5558)
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* Generate a warning message if sound files do not exist when trying to
play the user count to the conference. Use the new helper routine
sound_file_exists() for consistency.
* Put the new user into autoservice when playing user counts to the
conference.
* Check the return value of ast_bridge_impart().
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When r376428 was commited to re-order start up sequences to be more tolerant of
forking with thread primitives, a few items were changed that caused changes
in behavior on some distros. This includes:
* Not displaying the splash screen on a remote console.
* Displaying an error message on stderr when a remote console cannot connect
to a running instance of Asterisk.
In the first case, the splash screen was re-added (thanks to Michael L. Young).
In the second case, the various init.d scripts were modified to pipe stderr
to /dev/null, as the error message is useful - if you execute a remote
console or a remote console command execution and it fail, it should tell
you. Note that the error message was always present, it just failed to be
printed prior to r376428.
Much thanks to the folks who quickly reported this problem, provided solutions,
and promptly tested the various init.d scripts on a variety of distros.
(closes issue ASTERISK-20945)
Reported by: Warren Selby
Tested by: Michael L. Young, Jamuel Starkey, kaldemar, Danny Nicholas, mjordan
patches:
asterisk-20945-remote-intro-msg.diff uploaded by elguero (license 5026)
ASTERISK-20945-1.8-mjordan.diff uploaded by mjordan (license 6283)
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