Rusty Newton [Mon, 8 Apr 2013 23:34:59 +0000 (23:34 +0000)]
Modified the list of keys for the driver backends for sake of sample clarity
Added a line showing the mapping of "mysql" to res_config_mysql available in add-ons. We used "mysql" as an example driver key in the sample, but didn't show what module it mapped too. Also added a subtitle above the list of keys for driver backends.
Fix For Not Overriding The Default Settings In chan_sip
The initial report was that the "nat" setting in the [general] section was not
having any effect in overriding the default setting. Upon confirming that this
was happening and looking into what was causing this, it was discovered that
other default settings would not be overriden as well.
This patch works similar to what occurs in build_peer(). We create a temporary
ast_flags structure and using a mask, we override the default settings with
whatever is set in the [general] section.
In the bug report, the reporter who helped to test this patch noted that the
directmedia settings were being overriden properly as well as the nat settings.
(closes issue ASTERISK-21225)
Reported by: Alexandre Vezina
Tested by: Alexandre Vezina, Michael L. Young
Patches:
asterisk-21225-handle-options-default-prob_1.8_v4.diff.diff
Michael L. Young (license 5026)
In ASTERISK-20904, the focus was around the changes to NAT that took place in
Asterisk 11. Since the report stated that 1.8 was fine, we didn't take a look
at 1.8 at the time.
While working on ASTERISK-21225, I could see that 1.8 would benefit from having
some of those changes applied to it.
This patch does the following:
* The important part of this patch is that it sets the peer's flags earlier in
build_peer so that the code properly uses the peer's flags based on the peer's
configuration.
* constify req parameter in check_via()
* update realtime schemas under the contrib directory to handle properly the NAT
settings available in 1.8 as well as to handle the changes made in 11 to make
upgrading easier when installing newer versions of Asterisk
(closes issue ASTERISK-21243)
Reported by: Michael L. Young
Patches:
asterisk-20904-changes_for_1.8.diff Michael L. Young (license 5026)
The new inband_on_proceeding option causes Asterisk to assume inband audio
may be present when a PROCEEDING message is received.
Q.931 Section 5.1.2 says the network cannot assume that the CPE side has
attached to the B channel at this time without explicitly sending the
progress indicator ie informing the CPE side to attach to the B channel
for audio. However, some non-compliant ISDN switches send a PROCEEDING
without the progress indicator ie indicating inband audio is available and
assume that the CPE device has connected the media path for listening to
ringback and other messages.
ASTERISK-17834 which causes this issue was dealing with a non-compliant
network switch.
David M. Lee [Tue, 2 Apr 2013 17:33:53 +0000 (17:33 +0000)]
Fixed spurious rebuilds of func_version.
func_version.so was being rebuilt every time, because build.h was
changing every build, because of the cleantest dependency that was
added in r384410 to fix parallel make bugs.
Now build.h will only be created if it does not exist, which was the
original behavior of the Makefile.
David M. Lee [Mon, 1 Apr 2013 13:18:15 +0000 (13:18 +0000)]
Fix parallel make problems.
Occasionally, make -j would fail due to missing includes, or other
unusual errors.
This was due to the 'cleantest' target, which was designed to force a
make clean when some change in the code would cause the typical
depedency checking to fail. Several targets in the main Makefile did
not depend upon cleantest, hence would run in parallel to it. By
adding the dependency, make -j runs happily now.
Jonathan Rose [Fri, 29 Mar 2013 16:23:03 +0000 (16:23 +0000)]
app_voicemail: Add blank argument to externnotify if no context argument
At least one call to run_externnotify provides a NULL context parameter and
because the snprintf statement doesn't account for a NULL context parameter,
it simply writes '(null)' to the arguments string instead. This patch makes
it write two quotes back to back for that argument instead in the event of
a NULL context.
(closes issue ASTERISK-18207)
Reported by: Barry L. Kline
Patches:
modified from patch-20130306 uploaded by Karsten Wemheuer (License 5930)
Matthew Jordan [Wed, 27 Mar 2013 18:49:17 +0000 (18:49 +0000)]
Fix a file descriptor leak in off nominal path
While looking at the security vulnerability in ASTERISK-20967, Walter noticed
a file descriptor leak and some other issues in off nominal code paths. This
patch corrects them.
Note that this patch is not related to the vulnerability in ASTERISK-20967,
but the patch was placed on that issue.
Kinsey Moore [Wed, 27 Mar 2013 17:02:32 +0000 (17:02 +0000)]
Fix white noise on SRTP decryption
When res_rtp_asterisk.c was altered to avoid attempting to apply
unprotect algorithms to non-audio RTP packets, the test used was
incorrect. This caused the audio packets to not be decrypted and
resulted in loud white noise on the other endpoint (or both endpoints
depending on the call legs involved). The test now properly checks the
version field in the RTP header to ensure that RTP and RTCP are
decrypted while other types of packets are not.
(closes issue ASTERISK-21323)
Reported by: andrea
Tested by: Kinsey Moore, andrea, John Bigelow
Patches:
whitenoise_fix.diff uploaded by Kinsey Moore
Matthew Jordan [Wed, 27 Mar 2013 14:53:13 +0000 (14:53 +0000)]
AST-2013-003: Prevent username disclosure in SIP channel driver
When authenticating a SIP request with alwaysauthreject enabled, allowguest
disabled, and autocreatepeer disabled, Asterisk discloses whether a user
exists for INVITE, SUBSCRIBE, and REGISTER transactions in multiple ways. The
information is disclosed when:
* A "407 Proxy Authentication Required" response is sent instead of a
"401 Unauthorized" response
* The presence or absence of additional tags occurs at the end of "403
Forbidden" (such as "(Bad Auth)")
* A "401 Unauthorized" response is sent instead of "403 Forbidden" response
after a retransmission
* Retransmission are sent when a matching peer did not exist, but not when a
matching peer did exist.
This patch resolves these various vectors by ensuring that the responses sent
in all scenarios is the same, regardless of the presence of a matching peer.
This issue was reported by Walter Doekes, OSSO B.V. A substantial portion of
the testing and the solution to this problem was done by Walter as well - a
huge thanks to his tireless efforts in finding all the ways in which this
setting didn't work, providing automated tests, and working with Kinsey on
getting this fixed.
Matthew Jordan [Wed, 27 Mar 2013 14:35:11 +0000 (14:35 +0000)]
AST-2013-002: Prevent denial of service in HTTP server
AST-2012-014, fixed in January of this year, contained a fix for Asterisk's
HTTP server for a remotely-triggered crash. While the fix put in place fixed
the possibility for the crash to be triggered, a denial of service vector still
exists with that solution if an attacker sends one or more HTTP POST requests
with very large Content-Length values. This patch resolves this by capping
the Content-Length at 1024 bytes. Any attempt to send an HTTP POST with
Content-Length greater than this cap will not result in any memory allocation.
The POST will be responded to with an HTTP 413 "Request Entity Too Large"
response.
This issue was reported by Christoph Hebeisen of TELUS Security Labs
(closes issue ASTERISK-20967)
Reported by: Christoph Hebeisen
patches:
AST-2013-002-1.8.diff uploaded by mmichelson (License 5049)
AST-2013-002-10.diff uploaded by mmichelson (License 5049)
AST-2013-002-11.diff uploaded by mmichelson (License 5049)
Matthew Jordan [Tue, 26 Mar 2013 02:23:01 +0000 (02:23 +0000)]
Resolve deadlock between SIP registration and channel based functions
In r373424, several reentrancy problems in chan_sip were addressed. As a
result, the SIP channel driver is now properly locking the channel driver
private information in certain operations that it wasn't previously. This
exposed two latent problems either in register_verify or by functions called
by register_verify. This includes:
* Holding the private lock while calling sip_send_mwi_to_peer. This can create
a new sip_pvt via sip_alloc, which will obtain the channel container lock.
This is a locking inversion, as any channel related lock must be obtained
prior to obtaining the SIP channel technology private lock.
* Holding the privat elock while calling sip_poke_peer. In the same vein as
sip_send_mwi_to_peer, sip_poke_peer can create a new SIP private, causing
the same locking inversion.
Note that this locking inversion typically occured when CLI commands were run
while a SIP REGISTER request was being processed, as many CLI commands (such
as 'sip show channels', 'core show channels', etc.) have to obtain the channel
container lock.
(issue ASTERISK-21068)
Reported by: Nicolas Bouliane
(issue ASTERISK-20550)
Reported by: David Brillert
Matthew Jordan [Tue, 26 Mar 2013 01:50:28 +0000 (01:50 +0000)]
Resolve deadlock between pending CDR and batch CDR locks
r375757 attempted to resolve a race condition between multiple submissions of
CDRs while in batch mode from attempting to destroy the scheduled batch
submission by extending the batch CDR lock. Unfortunately, this causes a
deadlock between the pending CDR lock and the batch CDR lock. This patch
resolves the intent of r375757 by simply providing a new lock that protects
the scheduling of the batches. The original batch CDR lock is kept to protect
manipulation of the batch CDR settings, but has been placed such that it
is not held when the pending lock is held.
Thanks to Chase Venters for providing lock analysis on the issue.
Russell Bryant [Tue, 26 Mar 2013 01:32:27 +0000 (01:32 +0000)]
Fix multi-station answer race condition.
When an SLA trunk is ringing (inbound call on the trunk) Asterisk will
make outbound calls to the stations that have that trunk. If more than
one station answers the call at the same time, all channels other than
the first one to answer are left in a bad state. The channel gets
leaked, is not connected to anything, and there's no way to get rid of
it.
We now properly clean up these losing channels by hanging up on them.
Since they lost the race, as we process their answer, there is no
ringing trunk for them to answer.
Richard Mudgett [Mon, 25 Mar 2013 23:19:06 +0000 (23:19 +0000)]
Set the CALLERID(dnid-num-plan) for incoming ISDN calls.
The CALLEDTON channel variable is set for incoming ISDN calls to the lower
7 bits of the Q.931 type-of-number/numbering-plan octet. The
CALLERID(dnid-num-plan) should have the same value.
Kinsey Moore [Fri, 15 Mar 2013 12:49:59 +0000 (12:49 +0000)]
tcptls: Prevent unsupported options from being set
AMI, HTTP, and chan_sip all support TLS in some way, but none of them
support all the options that Asterisk's TLS core is capable of
interpreting. This prevents consumers of the TLS/SSL layer from setting
TLS/SSL options that they do not support.
This also gets tlsverifyclient closer to a working state by requesting
the client certificate when tlsverifyclient is set. Currently, there is
no consumer of main/tcptls.c in Asterisk that supports this feature and
so it can not be properly tested.
Review: https://reviewboard.asterisk.org/r/2370/ Reported-by: John Bigelow Patch-by: Kinsey Moore
(closes issue AST-1093)
Matthew Jordan [Fri, 15 Mar 2013 01:32:06 +0000 (01:32 +0000)]
When a session timer expires during a T.38 call, re-invite with correct SDP
When a session timer expires during a dialog that has re-negotiated to T.38
and Asterisk is the refresher, Asterisk will send a re-INVITE with an SDP
containing audio media only. This causes some hilarity with the poor fax
session under weigh.
This patch corrects that by sending T.38 parameters if we are in the middle of
a T.38 session.
Matthew Jordan [Fri, 15 Mar 2013 01:21:55 +0000 (01:21 +0000)]
Fix processing of call files when using KQueue on OS X
In certain situations, call files are not processed when using KQueue with
pbx_spool. Asterisk was sending an invalid timeout value when the spool
directory is empty, causing the call to kevent to error immediately. This
can create a tight loop, increasing the CPU load on the system.
Michael L. Young [Tue, 12 Mar 2013 21:15:52 +0000 (21:15 +0000)]
Fix Sorting Order For Parking Lots Stored In Static Realtime
When retrieving the parking lots from a MySQL database table, the current order
is "filename, cat_metric desc, var_metric asc, category". If there are multiple
parking lots with the same cat_metric but different categories, everything is
being sorted on cat_metric first resulting in errors when loading the parking
lots.
This patch fixes the problem by sorting on the category field first, then the
cat_metric field.
(closes issue ASTERISK-21035)
Reported by: Alex Epshteyn
Patches:
asterisk-21035-orderby.diff Michael L. Young (license 5026)
Matthew Jordan [Tue, 12 Mar 2013 16:20:30 +0000 (16:20 +0000)]
Include the Username field in SIP Registry events when Status is registered
In ASTERISK-17888, the AMI Registry event during SIP registrations was supposed
to include the Username field. Somehow, one of the events was missed. This
patch corrects that - the Username field should be included in all AMI Registry
events involving SIP registrations.
Matthew Jordan [Thu, 28 Feb 2013 17:09:25 +0000 (17:09 +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)
Matthew Jordan [Thu, 28 Feb 2013 16:40:31 +0000 (16:40 +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
Tzafrir Cohen [Tue, 26 Feb 2013 19:37:33 +0000 (19:37 +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 [Mon, 25 Feb 2013 12:48:52 +0000 (12:48 +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 22:59:11 +0000 (22:59 +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)
Matthew Jordan [Sun, 24 Feb 2013 16:25:08 +0000 (16:25 +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
Matthew Jordan [Sun, 24 Feb 2013 15:33:32 +0000 (15:33 +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.
Matthew Jordan [Thu, 21 Feb 2013 22:44:59 +0000 (22:44 +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.
Kevin Harwell [Tue, 19 Feb 2013 19:16:44 +0000 (19:16 +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
Walter Doekes [Mon, 18 Feb 2013 20:28:39 +0000 (20:28 +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.
Matthew Jordan [Thu, 14 Feb 2013 03:42:06 +0000 (03:42 +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 20:16:24 +0000 (20:16 +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)
Kevin Harwell [Mon, 11 Feb 2013 20:46:37 +0000 (20:46 +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.
David M. Lee [Wed, 6 Feb 2013 20:10:14 +0000 (20:10 +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.
David M. Lee [Thu, 31 Jan 2013 19:56:16 +0000 (19:56 +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/
Matthew Jordan [Wed, 30 Jan 2013 17:44:11 +0000 (17:44 +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)
Matthew Jordan [Wed, 30 Jan 2013 14:19:14 +0000 (14:19 +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.
(closes issue ASTERISK-20897)
Reported by: Eric Hill
Tested by: Eric Hill
patches:
-- uploaded by Eric Hill
Matthew Jordan [Wed, 30 Jan 2013 14:12:19 +0000 (14:12 +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
Richard Mudgett [Tue, 29 Jan 2013 17:22:22 +0000 (17:22 +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 17:05:18 +0000 (17:05 +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.
Matthew Jordan [Tue, 29 Jan 2013 02:02:32 +0000 (02:02 +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)
Sean Bright [Mon, 28 Jan 2013 21:06:01 +0000 (21:06 +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 01:52:39 +0000 (01:52 +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.
Jonathan Rose [Tue, 22 Jan 2013 18:21:04 +0000 (18:21 +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/
Matthew Jordan [Tue, 22 Jan 2013 14:43:05 +0000 (14:43 +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)
Matthew Jordan [Mon, 21 Jan 2013 20:19:54 +0000 (20:19 +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)
Kinsey Moore [Mon, 21 Jan 2013 18:27:58 +0000 (18:27 +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)
Matthew Jordan [Mon, 21 Jan 2013 04:05:29 +0000 (04:05 +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
Matthew Jordan [Fri, 18 Jan 2013 23:26:02 +0000 (23:26 +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.
(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)
David M. Lee [Fri, 18 Jan 2013 05:23:57 +0000 (05:23 +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)
Matthew Jordan [Thu, 17 Jan 2013 02:28:31 +0000 (02:28 +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)
Matthew Jordan [Wed, 16 Jan 2013 04:10:15 +0000 (04:10 +0000)]
Fix parsing SMSSRC for SMS messages
The parser for SMS messages would incorrectly parse out the from number.
The parsing would incorrectly start scanning for the from number at the
same index as the first double quote ("); this would inadvertently cause
it to treat the first double quote as the terminating double quote for
the from number as well.
The SMSSRC should now populate correctly.
(closes issue ASTERISK-16822)
Reported by: menschentier
Tested by: Jonas Falck
patches:
fixSMSSRC.patch uploaded by jonax (license 6320)
Matthew Jordan [Wed, 16 Jan 2013 00:10:37 +0000 (00:10 +0000)]
Set the INVALID_EXTEN channel variable when chan_misdn forces the 'i' extension
The chan_misdn channel driver will send a channel with an invalid destination
to the 'i' extension itself if said extension can be reached. It forgot,
however, to set the INVALID_EXTEN channel variable when it bounces the channel
to this extension. Dialplan writers everywhere moaned at yet another
inconsistency.
This is yet another example of why duplicating logic in multiple places results
in bugs that stick around in Jira for just under three years.
Yes: ASTERISK-15456 was created on January 18th, 2010. Patch committed on
January 15th, 2013. Ouch.
(closes issue ASTERISK-15456)
Reported by: Thomas Omerzu
patches:
chan_misdn_invalid.patch2 uploaded by Thomas Omerzu (license 5927)
Matthew Jordan [Tue, 15 Jan 2013 03:46:57 +0000 (03:46 +0000)]
Prevent crash in ConfBridge due to race condition when channels leave bridge
When a channel leaves a bridge, a race condition existed where the
bridge_channel's pvt structure would be accessed after it was disposed of.
This patch prevents that by setting the pointer to the pvt to NULL prior
to disposing of it.
Note that this patch is a backport from Asterisk 10. This particular race
condition was fixed as part of the larger code rework that occurred for that
release.
The solution to this problem was pointed out by Gunnar Harms in ASTERISK-16640.
David M. Lee [Mon, 14 Jan 2013 15:11:39 +0000 (15:11 +0000)]
Fix XML encoding of 'identity display' in NOTIFY messages, continued.
When r378933 was merged into 1.8, it should have also escaped
remote_display, since it will have the same XML encoding problem when
the caller/callee roles are reversed.
Matthew Jordan [Sun, 13 Jan 2013 21:15:06 +0000 (21:15 +0000)]
Reset RTP timestamp; sequence number on SSRC change
In r370252 for ASTERISK-18404, Asterisk's handling of RTP was modified to
better account for out of order RTP packets. This was accomplished by using the
RTP timestamp and sequence number to check for out of order packets. However,
when a SSRC change occurs, the timestamp and sequence number will no longer
have any relation to the previously received packets. The variables tracking
the timestamp and sequence number therefore have to be reset.
David M. Lee [Sat, 12 Jan 2013 06:26:34 +0000 (06:26 +0000)]
Fix XML encoding of 'identity display' in NOTIFY messages.
XML encoding in chan_sip is accomplished by naively building the XML
directly from strings. While this usually works, it fails to take into
account escaping the reserved characters in XML.
This patch adds an 'ast_xml_escape' function, which works similarly to
'ast_uri_encode'. This is used to properly escape the local_display
attribute in XML formatted NOTIFY messages.
Several things to note:
* The Right Thing(TM) to do would probably be to replace the
ast_build_string stuff with building an ast_xml_doc. That's a much
bigger change, and out of scope for the original ticket, so I
refrained myself.
* It is with great sadness that I wrote my own ast_xml_escape
function. There's one in libxml2, but it's knee-deep in
libxml2-ness, and not easily used to one-off escape a
string.
* I only escaped the string we know is causing problems
(local_display). At least some of the other strings are
URI-encoded, which should be XML safe. Rather than figuring out
what's safe and escaping what's not, it would be much cleaner to
simply build an ast_xml_doc for the messages and let the XML
library do the XML escaping. Like I said, that's out of scope.
Jonathan Rose [Fri, 4 Jan 2013 22:54:31 +0000 (22:54 +0000)]
res_srtp: Prevent a crash from occurring due to srtp_create failures in srtp_create
Under some circumstances, libsrtp's srtp_create function deallocates memory that
it wasn't initially responsible for allocating. Because we weren't initially
aware of this behavior, this memory was still used in spite of being unallocated
during the course of the srtp_unprotect function. A while back I made a patch
which would set this value to NULL, but that exposed a possible condition where
we would then try to check a member of the struct which would cause a segfault.
In order to address these problems, ast_srtp_unprotect will now set an error value
when it ends without a valid SRTP session which will result in the caller of
srtp_unprotect observing this error and hanging up the relevant channel instead of
trying to keep using the invalid session address.
Fix SIP Notify Messages To Have The Proper IP Address In The FROM Field
On a multihomed server when sending a NOTIFY message, we were not figuring out
which network should be used to contact the peer.
This patch fixes the problem by calling ast_sip_ouraddrfor() and then
build_via() so that our NOTIFY message contains the correct IP address.
Also, a debug message is being added to help follow the call-id changes that
occur. This was helpful for confirming that the IP address was set properly
since the call-id contains the IP address. It also will be helpful for
troubleshooting purposes when following a call in the debug logs.
(closes issue ASTERISK-20805)
Reported by: Bryan Hunt
Tested by: Bryan Hunt, Michael L. Young
Patches:
asterisk-20805-notify-ip-v2.diff uploaded by Michael L. Young (license 5026)
Fix Queue Log Reporting Every Call COMPLETECALLER With "h" Extension Present
When the "h" extension is present within the context of the queue, all calls
are being reported COMPLETECALLER even when the agent is hanging up the call.
This patch checks to see if the agent hung-up or not instead of only relying on
checking if the queue (caller) channel hung-up or not. It would appear that
having the h extension in the mix, the pbx goes to the h extension,
"hanging-up" the queue channel and triggering the reporting of COMPLETECALLER.
(closes issue ASTERISK-20743)
Reported by: call
Tested by: call, Michael L. Young
Patches:
asterisk-20743-q-cmplt-caller.diff
uploaded by Michael L. Young (license 5026)
Richard Mudgett [Thu, 3 Jan 2013 19:40:55 +0000 (19:40 +0000)]
chan_agent: Fix wrapup time wait response.
* Made agent_cont_sleep() and agent_ack_sleep() stop waiting if the wrapup
time expires. agent_cont_sleep() had tried but returned the wrong value
to stop waiting.
* Made agent_ack_sleep() take a struct agent_pvt pointer instead of a void
pointer for better type safety.
Kinsey Moore [Thu, 3 Jan 2013 18:35:38 +0000 (18:35 +0000)]
Add missing test event
This test event was missing from channel.c causing the dial_LS_options
test to fail intermittently because of a race condition where most code
paths emitted the test event but this one did not. The dial_LS_options
test should stop bouncing now.
Matthew Jordan [Wed, 2 Jan 2013 21:48:57 +0000 (21:48 +0000)]
Prevent crashes from occurring when reading from data sources with large values
When reading configuration data from an Asterisk .conf file or when pulling
data from an Asterisk RealTime backend, Asterisk was copying the data on the
stack for manipulation. Unfortunately, it is possible to read configuration
data or realtime data from some data source that provides a large blob of
characters. This could potentially cause a crash via a stack overflow.
This patch prevents large sets of data from being read from an ARA backend or
from an Asterisk conf file.
Richard Mudgett [Wed, 2 Jan 2013 21:08:15 +0000 (21:08 +0000)]
Fix AMI redirect action with two channels failing to redirect both channels.
The AMI redirect action can fail to redirect two channels that are bridged
together. There is a race between the AMI thread redirecting the two
channels and the bridge thread noticing that a channel is hungup from the
redirects.
* Made the bridge wait for both channels to be redirected before exiting.
* Made the AMI redirect check that all required headers are present before
proceeding with the redirection.
* Made the AMI redirect require that any supplied ExtraChannel exist
before proceeding. Previously the code fell back to a single channel
redirect operation.
(closes issue ASTERISK-18975)
Reported by: Ben Klang
Matthew Jordan [Wed, 2 Jan 2013 16:54:20 +0000 (16:54 +0000)]
Prevent exhaustion of system resources through exploitation of event cache
Asterisk maintains an internal cache for devices in the event subsystem. The
device state cache holds the state of each device known to Asterisk, such that
consumers of device state information can query for the last known state for
a particular device, even if it is not part of an active call. The concept of
a device in Asterisk can include entities that do not have a physical
representation. One way that this occurred was when anonymous calls are allowed
in Asterisk. A device was automatically created and stored in the cache for
each anonymous call that occurred; this was possible in the SIP and IAX2
channel drivers and through channel drivers that utilized the
res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif). These devices
are never removed from the system, allowing anonymous calls to potentially
exhaust a system's resources.
This patch changes the event cache subsystem and device state management to
no longer cache devices that are not associated with a physical entity.
(issue ASTERISK-20175)
Reported by: Russell Bryant, Leif Madsen, Joshua Colp
Tested by: kmoore
patches:
event-cachability-3.diff uploaded by jcolp (license 5000)
Matthew Jordan [Wed, 2 Jan 2013 15:16:10 +0000 (15:16 +0000)]
Resolve crashes due to large stack allocations when using TCP
Asterisk had several places where messages received over various network
transports may be copied in a single stack allocation. In the case of TCP,
since multiple packets in a stream may be concatenated together, this can
lead to large allocations that overflow the stack.
This patch modifies those portions of Asterisk using TCP to either
favor heap allocations or use an upper bound to ensure that the stack will not
overflow:
* For SIP, the allocation now has an upper limit
* For HTTP, the allocation is now a heap allocation instead of a stack
allocation
* For XMPP (in res_jabber), the allocation has been eliminated since it was
unnecesary.
Note that the HTTP portion of this issue was independently found by Brandon
Edwards of Exodus Intelligence.
(issue ASTERISK-20658)
Reported by: wdoekes, Brandon Edwards
Tested by: mmichelson, wdoekes
patches:
ASTERISK-20658_res_jabber.c.patch uploaded by mmichelson (license 5049)
issueA20658_http_postvars_use_malloc2.patch uploaded by wdoekes (license 5674)
issueA20658_limit_sip_packet_size3.patch uploaded by wdoekes (license 5674)
Kinsey Moore [Mon, 31 Dec 2012 14:41:37 +0000 (14:41 +0000)]
Ensure chan_sip rejects encrypted streams without crypto info
This ensures that Asterisk rejects encrypted media streams (RTP/SAVP
audio and video) that are missing cryptographic keys and ensures that
the incoming SDP is consistent with RFC4568 as far as having a crypto
attribute present for any SAVP streams.
Mark Michelson [Thu, 20 Dec 2012 20:26:47 +0000 (20:26 +0000)]
Adjust RTP instance's available_formats callback to return the correct type.
The RTP engine public function that gets the available formats expects a
format_t to be returned; however when calling into an RTP instance's
callback to get the available formats, the callback returned an int.
This never was noticed in Asterisk because the two RTP engines included
do not provide an available_formats callback.
This introduces an API change, and the proposal for this change was brought
up on the Asterisk developers mailing list [1]. There was no public objection
to this change, so it is now being put in.
projects / thirdparty / asterisk.git / log
