Jonathan Rose [Fri, 8 Mar 2013 20:16:43 +0000 (20:16 +0000)]
chan_sip: Update the via header when relaying SMS MESSAGE
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.
Matthew Jordan [Thu, 7 Mar 2013 17:57:08 +0000 (17:57 +0000)]
Let vm_mailbox_snapshot combine "Urgent" when no folder is specified
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.
Matthew Jordan [Thu, 7 Mar 2013 14:58:23 +0000 (14:58 +0000)]
Add a 'secret' probation strictrtp mode to handle delayed changes in RTP source
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
Kevin Harwell [Mon, 4 Mar 2013 20:03:09 +0000 (20:03 +0000)]
Confbridge CLI new record file name check.
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.
Fix / Clean Up Some Items To Handle The New auto_* NAT Options
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)
Joshua Colp [Thu, 28 Feb 2013 21:37:01 +0000 (21:37 +0000)]
Fix a bug with ICE and strictrtp where media could get dropped.
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.
Matthew Jordan [Thu, 28 Feb 2013 17:16:31 +0000 (17:16 +0000)]
Prevent deadlock in chan_iax2 when attempting to set caller ID
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
Matthew Jordan [Thu, 28 Feb 2013 16:52:34 +0000 (16:52 +0000)]
Let channels joining a MeetMe conference opt out of the denoiser
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
........
Merged revisions 382227 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Tzafrir Cohen [Tue, 26 Feb 2013 19:45:09 +0000 (19:45 +0000)]
Consider linux-gnuspe as linux-gnu
* 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
Matthew Jordan [Tue, 26 Feb 2013 15:35:05 +0000 (15:35 +0000)]
Clean up ConfBridge commands to account for wait_marked users
When ConfBridge was refactored to better handle the concept of marked,
wait_marked, and normal users co-existing in a conference (thereby implementing
a state machine for the conference), the wait_marked users were put into their
own list of conference participants, separate from the active users. This list
is used for wait_marked users when they are waiting in a conference but no
marked user has joined; normal users may have joined at this point however.
There are several AMI/CLI commands that affect conference users that were not
checking the wait_marked users list:
* CLI/AMI commands that mute/unmute a participant. In this case, wait_marked
users have to remain in their particular state and should not be affected -
however, the commands would return "Channel not found" as opposed to the
appropriate error condition.
* CLI/AMI commands that kick a participant. An admin should always be able to
kick a participant out of the conference.
This patch fixes both sets of commands, and cleans up the CLI commands slightly
by allowing them to complete a participant name (this was supposed to have been
added, but the function call was commented out and wasn't implemented).
Review: https://reviewboard.asterisk.org/r/2346/
(closes issue AST-1114)
Reported by: John Bigelow
Tested by: John Bigelow
Matthew Jordan [Tue, 26 Feb 2013 15:24:35 +0000 (15:24 +0000)]
Ensure that the default bridge/user profiles are always available
ConfBridge and Page require that there always be a default bridge and user
profile available. While properties of the default profiles can be overriden
in the configuration file, removing them can create situations where neither
application can function properly.
This patch ensures that if an administrator removes the profiles from the
confbridge.conf configuration file, the profiles are added upon load.
Documentation clarifying this has been added to the confbridge.conf.sample file.
Review: https://reviewboard.asterisk.org/r/2356/
(closes issue AST-1115)
Reported by: John Bigelow
Tested by: John Bigelow
Matthew Jordan [Mon, 25 Feb 2013 12:50:07 +0000 (12:50 +0000)]
Clean up use of va_end/va_args in res_config_mysql
There were several problems using variadic argument macros in res_config_mysql.
* Improper use of va_end. Multiple calls to va_end were possible resulting in
an unbalanced matching of va_start/va_end.
* Calls to va_arg after a possible encounter of a SENTINEL value.
Matthew Jordan [Sun, 24 Feb 2013 23:00:05 +0000 (23:00 +0000)]
Set the sin_family on the bind address socket during initialization
Somehow, chan_jingle has managed to operate for years without setting the
sin_family on its bindaddr socket. This patch properly sets the field during
initial module load to AF_INET.
Note that the patch on the issue was modified slightly to change the
initialization of the socket from allocation of a chan_jingle private to the
module initialization, as the bindaddr object (which is static) only needs to
have the address set once.
(closes issue ASTERISK-19341)
Reported by: andre valentin
patches:
0105-chan_jingle.patch uploaded by avalentin (License 6064)
........
Merged revisions 381975 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Sun, 24 Feb 2013 16:26:10 +0000 (16:26 +0000)]
Don't display the AMI ALL class authorization for users if they don't have it
When converting AMI class authorizations to a string representation, the
method always appends the ALL class authorization. This is especially
important for events, as they should always communicate that class
authorization - even if the event itself does not specify ALL as a class
authorization for itself. (Events have always assumed that the ALL class
authorization is implied when they are raised)
Unfortunately, this did mean that specifying a user with restricted class
authorizations would show up in the 'manager show user' CLI command as
having the ALL class authorization.
Rather then modifying the existing string manipulation function, this patch
adds a function that will only return a string if the field being compared
explicitly matches class authorization field it is being compared against.
This prevents ALL from being returned unless it is actually specified for
the user.
(closes issue ASTERISK-20397)
Reported by: Johan Wilfer
........
Merged revisions 381939 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Sun, 24 Feb 2013 15:37:26 +0000 (15:37 +0000)]
Make ParkAndAnnounce return to priority + 1 when return context is not defined
The ParkAndAnnounce application documentation for the optional return_context
parameter states the following:
return_context
The goto-style label to jump the call back into after timeout. Default
'priority+1'.
Unfortunately, the application was sending the channel back into the dialplan
at 'priority', which is the ParkAndAnnounce application call. This causes an
infinite loop of the channel constantly being parked, announced, timed out,
parked, announced, timed out... while fun, especially for those callers you
wish to drive to the end of madness, this was not the intent of the
application.
Michael L. Young [Fri, 22 Feb 2013 19:38:06 +0000 (19:38 +0000)]
Fix FastAGI To Properly Check For A Connection
When IPv6 support was added to FastAGI, the intent was to have the ability to
check all addresses resolved for a host since we might receive an IPv4 address
and an IPv6 address. The problem with the current code, is that, since we are
doing O_NONBLOCK, we get EINPROGRESS when calling ast_connect() but are ignoring
this instead of handling it. We break out of the loop and continue on. When we
later call ast_poll(), it succeeds but we never check if we have a connection or
not on the socket level. We then attempt to send data to the host address that
we think is setup and it fails. We then check the errno and see that we have
"connection refused" and then return with agi failed.
This patch does the following:
* Handles EINPROGRESS by creating the function handle_connection()
- ast_poll() was moved into this function
- This function checks the results of the connection on the socket level after
calling ast_poll()
* Continues to the next address if the above fails to create a connection
* Once all addresses resolved are tried and we still are unable to establish a
connection, then we return that the FastAGI call failed
(closes issue ASTERISK-21065)
Reported by: Jeremy Kister
Tested by: Jeremy Kister, Michael L. Young
Patches:
asterisk-21065_poll_correctly_v4.diff Michael L. Young (license 5026)
Matthew Jordan [Thu, 21 Feb 2013 22:48:14 +0000 (22:48 +0000)]
Properly detect launchd
Asterisk was a little too pro-active in claiming that it found launchd. On
systems without launchd - such as FreeBSD - this resulted in certain items
in Asterisk that conflict with launchd to not be selectable, such as
res_timing_kqueue.
Matthew Jordan [Wed, 20 Feb 2013 19:14:32 +0000 (19:14 +0000)]
Let vm_mailbox_snapshot_create's combine option apply to "Urgent" as well
The vm_mailbox_snapshot_create function has an option that combines the
contents of INBOX and Old into a single snapshot. The intent of this is that
both 'new' messages and 'deleted' messages are given in a single snapshot, as
some applications prefer this view of the voicemail world. Unfortunately, the
initial implementation ignored the "Urgent" folder. The "Urgent" folder is a
pseudo-INBOX, in that new messages left with the 'U' flag will be placed in
that folder as opposed to INBOX. Thus, the option failed the intent with which
it was added.
This patch makes it so that the "Urgent" folder is included in the snapshot
when that option is used.
Kevin Harwell [Tue, 19 Feb 2013 19:44:58 +0000 (19:44 +0000)]
Write the correct callid to the data1 field in queue_log for transfer events.
The incorrect callid was being written to the "data1" field in queue_log table
for transfer events. The callid of the queue was being written instead of the
transfer target's callid. This now gets the correct "transfer to" number and
places that in the "data1" field of the queue_log table when a transfer event
is triggered.
(closes issue ASTERISK-19960)
Reported by: vladimir shmagin
........
Merged revisions 381770 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Kevin Harwell [Tue, 19 Feb 2013 16:21:18 +0000 (16:21 +0000)]
Confbridge channels staying active when all participants leave.
If you started/stopped recording of a conference multiple times channels
would remain active even when all participants left the conference. This
was due to the fact that a reference to the confbridge was being added
every time a start record command was issued, but when the recording was
stopped there was no matching de-reference thus keeping the conference alive.
Made sure only a single reference is added for the record thread no matter how
many times recording is started/stopped. A de-reference is issued upon thread
ending.
Note, this issue is being fixed under AST-1088 since it relates to it and
should have been corrected along with those modifications.
Kevin Harwell [Mon, 18 Feb 2013 22:21:10 +0000 (22:21 +0000)]
Fixed Confbridge file recording deadlock and appending.
A deadlock occurred after starting/stopping and then restarting a confbridge
recording. Upon starting a recording a record thread is created that holds a
lock until just before exiting. Stopping the recording does not stop/exit the
thread or release the lock. The thread waits until recording begins again.
Starting a stopped recording signals the thread to continue and start recording
again. However restarting the recording also created another record thread
resulting in a deadlock. The fix was to make sure the record thread was only
created once.
Also it was noted that filenames for the recordings were being concatenated for
each start/stop. This was fixed by creating a new file for each conference
session and appending the actual recorded data within the file (e.g. passing
the 'a' option to MixMonitor).
(issue AST-1088)
Reported by: John Bigelow
Review: http://reviewboard.digium.internal/r/374/
Walter Doekes [Mon, 18 Feb 2013 20:30:32 +0000 (20:30 +0000)]
Remove "registertrying" and add "rtp_engine" from/to sip.conf.sample
The "registertrying" option was removed in r343220. The "rtp_engine"
option was added in r186078 but erroneously named "engine" in the sample.
Note that there is no global sip setting for a different engine.
........
Merged revisions 381668 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Jonathan Rose [Mon, 18 Feb 2013 19:43:10 +0000 (19:43 +0000)]
PRESENCE_STATE: Provide better documentation for the 'e' option.
Notes that the 'e' option actually decodes data when used as a write function
such as with the SET application while it encodes data when used to read.
Matthew Jordan [Sat, 16 Feb 2013 16:22:37 +0000 (16:22 +0000)]
Don't send presencestate information if the state is invalid
Previously, presencestate information was sent whenever the state was not
NOT_SET. When r381594 actually returned INVALID presence state in all the
places it was supposed to, it caused chan_sip to start adding presence
state information to NOTIFY requests that it previously would not have
added. chan_sip shouldn't be adding presence state information when the
provider is in an invalid state; users can't set the state to invalid and
an invalid state always implies that the provider is in an error condition.
Matthew Jordan [Fri, 15 Feb 2013 23:23:49 +0000 (23:23 +0000)]
Fix crash in PresenceState AMI action when specifying an invalid provider
This patch fixes a crash in Asterisk that could be caused by using the
PresenceState AMI action while providing an invalid provider. This patch
also adds some additional warnings when a user attempts to provide the
PresenceState action with invalid data, and removes some NOTICE statements
that were still lurking in the code from testing.
(closes issue AST-1084)
Reported by: John Bigelow
Tested by: John Bigelow
Mark Michelson [Fri, 15 Feb 2013 18:42:02 +0000 (18:42 +0000)]
Fix a crash that occurred when a BYE was received on a replaced dialog.
Reference counting for the channel and its tech_pvt got messed up at
some point between 1.8 and 11. The result was that if a BYE for a dialog
that had been replaced (via an INVITE with Replaces) was received, Asterisk
would crash due to trying to access data on a channel that was no longer there.
The fix I introduced is to remove code that both unrefs the sip_pvt and sets
the channel's tech_pvt to NULL when an INVITE with Replaces is handled. This
way when a BYE is received, the tech_pvt will be non-NULL and so the BYE can
be processed and not cause a crash.
(closes issue ASTERISK-20929)
reported by Kristopher Lalletti
patches:
ASTERISK-20929.patch uploaded by Mark Michelson (License #5049)
Kevin Harwell [Fri, 15 Feb 2013 17:17:27 +0000 (17:17 +0000)]
Stopped spamming of debug messages during attended transfer.
While autoservice is running and servicing a channel the callid is being stored
and removed in the thread's local storage for each iteration of the thread loop.
If debug was set to a sufficient level the log file would be spammed with callid
thread local storage debug messages.
Added a new function that checks to see if the callid to be stored is different
than what is already contained (if anything). If it is different then
store/replace and log, otherwise just leave as is. Also made it so all logging
of debug messages pertaining to the callid thread storage outputs only when
TEST_FRAMEWORK is defined.
Jonathan Rose [Fri, 15 Feb 2013 17:12:20 +0000 (17:12 +0000)]
chan_sip: Use video and text crypto attributes to append RTP profiles to SDP
Some bad copy/pasting resulted in using the audio crypto attribute for both
text and video RTP. Also the audio crypto isn't set until after these, so it
was really just bad all around.
(closes ASTERISK-20905)
Reported by: Kristopher Lalletti
patches:
rtp_crypto_video_text.diff uploaded by Jonathan Rose (license 6182)
Matthew Jordan [Thu, 14 Feb 2013 03:48:39 +0000 (03:48 +0000)]
Don't throw a spurious error when using DBdeltree
The function call ast_db_deltree returns the number of row deleted, or a
negative number if it failed. DBdeltree was treating any non-zero return
as an error, causing a spurious verbose error message to be displayed.
This patch handles the return code of ast_db_deltree correctly.
Mark Michelson [Tue, 12 Feb 2013 20:31:52 +0000 (20:31 +0000)]
Do not allow native RTP bridging if packetization of media streams differs.
The RTP engine will no longer allow for local and remote native RTP bridges
if packetization of streams differs. Allowing native bridging in this scenario
has been known to cause FAX failures.
(closes ASTERISK-20650)
Reported by: Maciej Krajewski
Patches:
ASTERISK-20659.patch uploaded by Mark Michelson (License #5049)
Kinsey Moore [Tue, 12 Feb 2013 20:16:45 +0000 (20:16 +0000)]
Fix some more REF_DEBUG-related build errors
When sip_ref_peer and sip_unref_peer were exported to be usable in
channels/sip/security_events.c, modifications to those functions when
building under REF_DEBUG were not taken into account. This change
moves the necessary defines into sip.h to make them accessible to
other parts of chan_sip that need them.
Kevin Harwell [Mon, 11 Feb 2013 20:55:04 +0000 (20:55 +0000)]
Properly load say.conf upon reload of module app_playback.
If say.conf did not exists prior to originally loading module app_playback it
would not load on subsequent reloads of the module once it had been created.
This occurred because upon reload of the app_playback module it would only
load a new configuration if an old one had previously existed. This fix simply
removed the association between checking if an old configuration existed and
the loading of the new one.
Matthew Jordan [Mon, 11 Feb 2013 15:03:40 +0000 (15:03 +0000)]
Fix crash in res_xmpp when deleting pubsub node from CLI
An error existed in res_xmpp where it would attempt to delete attributes from
a node that itself was also deleted. Per the iksemel documentation, attributes
added using iks_insert are copied to the parent node's stack, and will be
reclaimed when that node is itself destroyed.
David M. Lee [Wed, 6 Feb 2013 20:14:32 +0000 (20:14 +0000)]
Fixed failing test from r380696.
When I added my extensive suite of session timer unit tests, apparently one of
them was failing and I never noticed. If neither Min-SE nor Session-Expires is
set in the header, it was responding with a Session-Expires of the global
maxmimum instead of the configured max for the endpoint.
(issue ASTERISK-20787)
........
Merged revisions 380973 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Damien Wedhorn [Wed, 6 Feb 2013 08:42:09 +0000 (08:42 +0000)]
Fix reload skinny with active devices.
Patch ensures that d->activeline and l->activesub are moved over to the
new device and line so that on callend the appropriate subs can be found
to complete hangup before device resets.
Damien Wedhorn [Wed, 6 Feb 2013 07:00:39 +0000 (07:00 +0000)]
Reset skinny vmexten on reload.
Make skinny reset vmexten '\0' on reload to ensure that
it is set to '\0' if the appropriate item is removed/commented in
skinny.conf. part of ASTERISK-21037
Reported by: snuffy
Tested by: snuffy, myself
Patches:
part of immed_dial_fix.diff uploaded by snuffy (license 5024)
Richard Mudgett [Tue, 5 Feb 2013 19:09:40 +0000 (19:09 +0000)]
app_page and app_confbridge: Fix custom announcement on entering conference.
The Page and ConfBridge custom announcement did not play when users
entered the conference.
* Fix the CONFBRIDGE(user,announcement) file not getting played. The code
to do this got removed accidentally when the ConfBridge code was
restructured to be more state machine like.
* Fixed play_prompt_to_user() doxygen comments.
* Fixed the Page A(x) and n options for the caller. The caller never
played the announcement file and totally ignored the n option. The code
to do this was lost when the application was converted to use ConfBridge.
* Factored out setup_profile_bridge(), setup_profile_paged(), and
setup_profile_caller() routines to setup ConfBridge profiles. Made each
profile setup routine use the default template if one has not already been
setup by dialplan.
(closes issue ASTERISK-20990)
Reported by: Jeremy Kister
Tested by: rmudgett
Richard Mudgett [Tue, 5 Feb 2013 18:49:32 +0000 (18:49 +0000)]
app_confbridge: Fix error messages on exiting conference.
A marked user ending a conference with only end_marked users generates
error messages:
ERROR[0000][C-00000000]: confbridge/conf_state.c:47 conf_invalid_event_fn: Invalid event for confbridge user ''
* The MULTI_MARKED state was doing too much when it was kicking out the
end_marked users from the conference. The kicked out users will clean up
after themselves when they exit the conference.
(closes issue ASTERISK-20991)
Reported by: Jeremy Kister
Tested by: rmudgett
David M. Lee [Thu, 31 Jan 2013 20:10:23 +0000 (20:10 +0000)]
Process session timers, even if Session-Expires header is missing
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/
........
Merged revisions 380696 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Wed, 30 Jan 2013 17:46:52 +0000 (17:46 +0000)]
Support building Asterisk for Raspberry Pi/Raspbian with hard-float support
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)
........
Merged revisions 380520 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Wed, 30 Jan 2013 15:07:59 +0000 (15:07 +0000)]
Unregister SIP provider API if module load is declined
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.
Matthew Jordan [Wed, 30 Jan 2013 14:20:47 +0000 (14:20 +0000)]
Perform case insensitive comparisons for T.38 attributes
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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Merged revisions 380458 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Wed, 30 Jan 2013 14:15:27 +0000 (14:15 +0000)]
Fix memory leak in res_calendar_icalendar
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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Merged revisions 380451 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Richard Mudgett [Tue, 29 Jan 2013 17:54:17 +0000 (17:54 +0000)]
chan_agent: Prevent multiple channels from logging in as the same agent.
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.
David M. Lee [Tue, 29 Jan 2013 17:14:51 +0000 (17:14 +0000)]
Corrected crypto tag in SDP ANSWER for SRTP. (again)
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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Merged revisions 380347 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Jonathan Rose [Tue, 29 Jan 2013 17:05:23 +0000 (17:05 +0000)]
call_parking: Make sure fallbacks are used when lacking a flat channel exten
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/
Matthew Jordan [Tue, 29 Jan 2013 14:45:46 +0000 (14:45 +0000)]
Ensure that a declined media stream is terminated with a '\r\n'
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
Matthew Jordan [Tue, 29 Jan 2013 02:09:51 +0000 (02:09 +0000)]
Update configure script to be compatible with ptlib 2.10.9
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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Merged revisions 380297 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Sean Bright [Mon, 28 Jan 2013 21:08:04 +0000 (21:08 +0000)]
Correct the number of available call numbers in IAX2.
There is currently an edge case where call number 32768 might be allocated for
a call, even though the IAX2 protocol requires call numbers be only 15 bits.
This resulted in some unpredictable behavior when call number 32678 is chosen.
This patch was mostly written by Richard Mudgett via ReviewBoard. I'm just
committing it.
Russell Bryant [Mon, 28 Jan 2013 01:57:26 +0000 (01:57 +0000)]
Change cleanup ordering in filestream destructor.
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.
Michael L. Young [Sun, 27 Jan 2013 20:31:39 +0000 (20:31 +0000)]
Fix Some Configured Conference Bridge Sounds Not Being Set
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)
David M. Lee [Thu, 24 Jan 2013 16:39:33 +0000 (16:39 +0000)]
Corrected crypto tag in SDP ANSWER for SRTP.
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)
Matthew Jordan [Thu, 24 Jan 2013 04:01:27 +0000 (04:01 +0000)]
Correct documentation for ConfbridgeList AMI action
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.
Jonathan Rose [Tue, 22 Jan 2013 19:07:42 +0000 (19:07 +0000)]
app_meetme: Use new prompts for administrator menu
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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Merged revisions 379885 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Tue, 22 Jan 2013 14:51:54 +0000 (14:51 +0000)]
Fix station ringback; trunk hangup issues in SLA
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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Merged revisions 379825 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Richard Mudgett [Tue, 22 Jan 2013 00:35:34 +0000 (00:35 +0000)]
confbridge: Minor fixes playing user counts to the conference.
* 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.
Matthew Jordan [Mon, 21 Jan 2013 20:40:13 +0000 (20:40 +0000)]
Update init.d scripts to handle stderr; readd splash screen for remote consoles
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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Merged revisions 379760 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 379777 from http://svn.asterisk.org/svn/asterisk/branches/10
Kinsey Moore [Mon, 21 Jan 2013 18:33:12 +0000 (18:33 +0000)]
Prevent segfault for interpolated iLBC frames
When iLBC is being used with a jitter buffer and the jb has to
interpolate frames, it generates frames with a null pointer and a
non-zero datalen. This is now handled properly.
(closes issue ASTERISK-20914)
Reported By: John McEleney
Patches:
ASTERISK-20914-1.8.diff uploaded by Matt Jordan (license 6283)
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Merged revisions 379718 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Damien Wedhorn [Mon, 21 Jan 2013 06:27:24 +0000 (06:27 +0000)]
Fix device call logging issues in skinny
Skinny device call logging (ie missed, place and received calls) has issues
because the incorrect sequence of callstates is/can be sent to the device.
This patch removes some extra callstate updates driven by forces external
to skinny and ensures the needed intermediary callstate messages are sent.
Matthew Jordan [Mon, 21 Jan 2013 04:07:05 +0000 (04:07 +0000)]
Fix crash in app_minivm when mime encoding string
An incorrect string initializations was left in ast_str_encode_mime from the
patch that converted string manipulations to use ast_str strings (r191140).
The string initialization causes a crash when ast_str_set is called on
the string later on in the function.
(closes issue ASTERISK-18697)
Reported by: Chris Boot
patches:
minivm-null-pointer-dereference-fix.patch uploaded by bootc (license 6309)
(issue ASTERISK-20854)
Reported by: Chris Warr
Tested by: Chris Warr
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Merged revisions 379608 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Sat, 19 Jan 2013 00:17:53 +0000 (00:17 +0000)]
Fix astcanary startup problem due to wrong pid value from before daemon call
When Asterisk forks itself into the background via a call to daemon, it must
re-set the pid value of the new process. Otherwise, astcanary gets the pid
value of the process before the fork, which prevents it from running. Asterisk
eventually starts lowering its priority, as it can no longer communicate
with the proverbial canary in the coal mine.
This patch ensures that the correct process identifier is used by astcanary.
Note that this is getting committed to 10 as a regression fix.
(closes issue ASTERISK-20947)
Reported by: Jakob Hirsch
Tested by: mjordan
patches:
asterisk-10.12.0.astcanary_ppid.diff uploaded by Jakob Hirsch (license 6113)
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Merged revisions 379509 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 379510 from http://svn.asterisk.org/svn/asterisk/branches/10
Kinsey Moore [Fri, 18 Jan 2013 21:46:58 +0000 (21:46 +0000)]
Fix regression in Confbridge user count
When the restructuring work got committed to Confbridge in r375470 to
fix many open issues, it caused a regression in the reported count of
users when conference information was requested via CLI or manager.
This corrects the user count and user information displayed when
listing conference information from the CLI and manager.
(closes issue ASTERISK-20938)
Reported By: Timo Teras
Patches:
confbridge-list.patch uploaded by Timo Teras (license 5409)
Jonathan Rose [Fri, 18 Jan 2013 18:13:58 +0000 (18:13 +0000)]
app_voicemail: Improve msg_id handling
app_voicemail will no longer issue error messages when it retrieves an msg_id
with a NULL value from realtime and will instead simply populate the msg_id
field with a newly generated msg_id. In addition, this patch changes the way
msg_ids are generated to eliminate certain causes of duplicate IDs appearing
within a single system. In addition, when messages are copied, they will now
receive a new msg_id.
(closes issue ASTERISK-20717)
Reported by: Alec Davis
Review: https://reviewboard.asterisk.org/r/2220/
David M. Lee [Fri, 18 Jan 2013 05:26:56 +0000 (05:26 +0000)]
Fix Record-Route parsing for large headers.
Record-Route parsing copied the header into a char[256] array, which can
be a problem if the header is longer than that. This patch parses the
header in place, without the copy, avoiding the issue.
In addition to the original patch, I added a unit test for the new
get_in_brackets_const function.
(closes issue ASTERISK-20837)
Reported by: Corey Farrell
Patches:
chan_sip-build_route-optimized-rev1.patch uploaded by Corey Farrell (license 5909)
(with minor changes by dlee)
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Merged revisions 379392 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Thu, 17 Jan 2013 02:30:29 +0000 (02:30 +0000)]
Fix issue where chan_mobile fails to bind to first available port
Per the bluez API, in order to bind to the first available port, the rc_channel
field of the socket addressing structure used to bind the socket should be set
to 0. Previously, Asterisk had set the rc_channel field set to 1, causing it
to connect to whatever happens to be on port 1.
We could probably not explicitly set rc_channel to 0 since we memset the struct
earlier, but explicitly setting it will hopefully prevent someone from coming
in and setting it to some explicit port in the future.
(closes issue ASTERISK-16357)
Reported by: challado
Tested by: Alexander Heinz, Nikolay Ilduganov, benjamin, eliafino, David van Geyn
patches:
ASTERISK-16357.diff uploaded by Nikolay Ilduganov (license 6253)
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Merged revisions 379342 from http://svn.asterisk.org/svn/asterisk/branches/1.8
projects / thirdparty / asterisk.git / log
