RFC3261 Section 8.1.1.5. The sequence number value MUST be expressible as a 32-bit unsigned integer
* fix: use %u instead of %d when dealing with CSeq numbers - to remove possibility of -ve numbers.
* fix: change all uses of seqno and friends (ocseq icseq) from 'int' or 'unsigned int' to uint32_t.
Summary of CSeq numbers.
An initial CSeq number must be less than 2^31
A CSeq number can increase in value up to 2^32-1
An incrementing CSeq number must not wrap around to 0.
Tested with Asterisk 1.8.8.2 with Grandstream phones.
Kevin P. Fleming [Mon, 30 Jan 2012 12:48:31 +0000 (12:48 +0000)]
Clarify log WARNING message when port-zero SDP 'm' lines received.
Previously, if an m-line in an SDP offer or answer had a port number of zero,
that line was skipped, and resulted in an 'Unsupported SDP media type...'
warning message. This was misleading, as the media type was not unsupported,
but was ignored because the m-line indicated that the media stream had been
rejected (in an answer) or was not going to be used (in an offer).
........
Merged revisions 353260 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Russell Bryant [Sun, 29 Jan 2012 02:44:24 +0000 (02:44 +0000)]
Find even more network interfaces.
The previous change made the code look for emN and pciN in addition to what
it did originally, which was search for ethN. However, it needed to be looking
for pciN#N, so that's what it does now.
This also moves the memset() to be before every ioctl().
........
Merged revisions 353175 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Kevin P. Fleming [Sat, 28 Jan 2012 14:51:29 +0000 (14:51 +0000)]
Add 'L16-256' MIME subtype alias for slin16.
Asterisk has supported the 'L16' MIME subtype for 16kHz signed linear (PCM)
audio for quite some time, but some endpoints refer to it as 'L16-256'. This
commit adds this as an alias for the existing format.
........
Merged revisions 353126 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Russell Bryant [Sat, 28 Jan 2012 04:27:55 +0000 (04:27 +0000)]
Update ast_set_default_eid() to find more network interfaces.
As of Fedora 15, ethN is not the name of ethernet interfaces. The names
are emN or pciN. Update some code that searched for interfaces named
ethN to look for the new names, as well. For more information about why
this change was made, see this page:
http://domsch.com/blog/?p=455
........
Merged revisions 353077 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Jonathan Rose [Fri, 27 Jan 2012 19:19:46 +0000 (19:19 +0000)]
Make failed PauseMonitor and UnpauseMonitor with no valid channel not close AMI session.
I also went ahead and took a little time to make sure that the manager value
AMI_SUCCESS was used instead of just return 0 being thrown around everywhere since that's
how we handle this stuff these days.
rfc4235 - Section 4.1: Versions MUST be representable using a non-negative 32 bit integer.
If a BLF subscription exists for long enough, using %d may print negative version numbers.
Unlikely, as 2^32 at 1 update per second is ~137 years, or half that before the versions number started going negative.
Tested with Asterisk 1.8.8.2 with Grandstream phones.
Jonathan Rose [Thu, 26 Jan 2012 19:07:01 +0000 (19:07 +0000)]
Copy amaflags to sip_pvt from peer during create_addr_from_peer
For whatever reason, we don't have a single function for copying data like this
from SIP peers to the SIP pvt. This patch adds the copying of amaflags to the
sip_pvt, but it would probably be worth discussing this function along with
the others that essentially just copy some amount of data from a peer to a
private.
(Closes issue ASTERISK-19029)
Reported by: Matt Lehner
........
Merged revisions 352755 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Kevin P. Fleming [Wed, 25 Jan 2012 21:18:22 +0000 (21:18 +0000)]
Avoid unnecessary rebuilds of main/test.c.
main/test.c includes "asterisk/version.h", when it should include
"asterisk/ast_version.h" instead (and it should use the ast_get_version()
and ast_get_version_num() functions). This commit modifies it to extract
the Asterisk version information using the proper APIs, and as a result means
that main/test.c no longer needs to be rebuilt when a Subversion checkout
is updated or modified.
........
Merged revisions 352612 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Richard Mudgett [Wed, 25 Jan 2012 17:16:22 +0000 (17:16 +0000)]
Fixes for sending SIP MESSAGE outside of calls.
* Fix authenticate MESSAGE losing custom headers added by the MESSAGE_DATA
function in the authorization attempt.
* Pass up better From header contents for SIP to use. Now is in the
"display-name" <URI> format expected by MessageSend. (Note that this is a
behavior change that could concievably affect some people.)
* Block user from adding standard headers that are added automatically.
(To, From,...)
* Allow the user to override the Content-Type header contents sent by
MessageSend.
* Decrement Max-Forwards header if the user transferred it from an
incoming message.
* Expand SIP short header names so the dialplan and other code only has to
deal with the full names.
* Documents what SIP expects in the MessageSend(from) parameter.
(closes issue ASTERISK-18992)
Reported by: Yuri
(closes issue ASTERISK-18917)
Reported by: Shaun Clark
Kevin P. Fleming [Wed, 25 Jan 2012 16:54:21 +0000 (16:54 +0000)]
Eliminate unnecessary rebuilds of main/format*.c.
These files have no need to include "asterisk/version.h", and doing so forces
them to be rebuilt each time a Subversion checkout moves between 'modified'
and 'unmodified' states.
Jonathan Rose [Wed, 25 Jan 2012 16:41:29 +0000 (16:41 +0000)]
Redocuments sip types peer, user, friend in sip.conf.sample
There was faulty information in the sample config describing user as a synonym for friend
so it has been changed to better elaborate on the differences between the three entity
types.
Jonathan Rose [Tue, 24 Jan 2012 20:35:38 +0000 (20:35 +0000)]
Set core sounds version to 1.4.22.
Now that we have the right license for the Russian 1.4.22 sounds as well as the sounds
for the Australian English 1.4.22 sounds, we can finally set the sounds to use 1.4.22!
(closes issue ASTERISK-18978)
Reported by: Cameron Twomey
Patches:
confbridge.tar.001 uploaded by Cameron Twomey
confbridge.tar.002 uploaded by Cameron Twomey
........
Merged revisions 352367 from http://svn.asterisk.org/svn/asterisk/branches/1.8
While the FAXOPT function could be used to set the modem capabilities, the
input to that function was not being applied correctly to the spandsp layer.
This patch applies the current model capabilities before starting the spandsp
layer.
(closes issue: ASTERISK-16409)
Reported by: Kristijan Vrban
Tested by: Matt Jordan, Matthew Nicholson
Patches:
spandsp-modems-1.8.diff uploaded by mnicholson (license 5081)
spandsp-modems-10.diff uploaded by mnicholson (license 5081)
........
Merged revisions 352144 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Mark Michelson [Sat, 21 Jan 2012 00:06:21 +0000 (00:06 +0000)]
Fix RTP reference leak.
If a blind transfer were initiated using a REFER without a prior
reINVITE to place the call on hold, AND if Asterisk were sending
RTCP reports, then there was a reference for the RTP instance
of the transferer.
This fixes the issue by merging two similar but slightly conflicting
sections of code into a single area. It also adds a stop_media_flows()
call in the case that the transferer's UA never sends a BYE to us
like it is supposed to.
Matthew Jordan [Fri, 20 Jan 2012 15:54:32 +0000 (15:54 +0000)]
Remove unused variable 'tmp' from helpfun in ilbc codec
gcc version 4.6.2 caught an unused variable in the ilbc codec
library. This would prevent compilation with --enable-dev-mode;
variable removed.
........
Merged revisions 351760 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Richard Mudgett [Thu, 19 Jan 2012 23:25:05 +0000 (23:25 +0000)]
Misc minor fixes in reqresp_parser.c and chan_sip.c.
* Fix corner cases in get_calleridname() parsing and ensure that the
output buffer is nul terminated.
* Make get_calleridname() truncate the name it parses if the given buffer
is too small rather than abandoning the parse and not returning anything
for the name. Adjusted get_calleridname_test() unit test to handle the
truncation change.
* Fix get_in_brackets_test() unit test to check the results of
get_in_brackets() correctly.
* Fix parse_name_andor_addr() to not return the address of a local buffer.
This function is currently not used.
* Fix potential NULL pointer dereference in sip_sendtext().
* No need to memset(calleridname) in check_user_full() or tmp_name in
get_name_and_number() because get_calleridname() ensures that it is nul
terminated.
* Reply with an accurate response if get_msg_text() fails in
receive_message(). This is academic in v1.8 because get_msg_text() can
never fail.
........
Merged revisions 351618 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Kinsey Moore [Thu, 19 Jan 2012 22:43:35 +0000 (22:43 +0000)]
Correct output of RTCP jitter statistics in SR and RR reports
Change the RTCP RR and SR generation code to convert Asterisk's internal jitter
statistics to be represented in RTP timestamp units based on the rate of the
codec in use instead of in seconds.
(closes issue ASTERISK-14530)
........
Merged revisions 351611 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Jonathan Rose [Thu, 19 Jan 2012 21:47:22 +0000 (21:47 +0000)]
Eliminates doubling the :port part of SIP Notify Message-Account headers.
This patch prevents the domain string from getting mangled during the initreqprep
step by moving the initialization to before its immediate use. It also documents
this pitfall for the ast_sockaddr_stringify functions.
Matthew Jordan [Wed, 18 Jan 2012 21:05:35 +0000 (21:05 +0000)]
Include iLBC source code for distribution with Asterisk
This patch includes the iLBC source code for distribution with Asterisk.
Clarification regarding the iLBC source code was provided by Google, and
the appropriate licenses have been included in the codecs/ilbc folder.
Stefan Schmidt [Wed, 18 Jan 2012 15:57:49 +0000 (15:57 +0000)]
The get_pai function in chan_sip.c didn't recognized a proper callerid name and
number from a P-Asserted-Identity cause the header parsing logic was wrong.
Changing the parsing functions to the sip header parsing APIs in
reqresp_parser.h solves this problem.
Review: https://reviewboard.asterisk.org/r/1673
Reviewed by: wdoekes2 and Mark Michelson
........
Merged revisions 351396 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Jonathan Rose [Tue, 17 Jan 2012 17:08:33 +0000 (17:08 +0000)]
Adds pjmedia probation concepts to res_rtp_asterisk's learning mode.
In order to better handle RTP sources with strictrtp enabled (which is now default in 10)
using the learning mode to figure out new sources when they change is handled by checking
for a number of consecutive (by sequence number) packets received to an rtp struct
based on a new configurable value called 'probation'. Also, during learning mode instead
of liberally accepting all packets received, we now reject packets until a clear source
has been determined.
Mark Michelson [Tue, 17 Jan 2012 16:54:31 +0000 (16:54 +0000)]
Use built-in parsing functions for Contact and Record-Route headers.
If a Contact or a Record-Route header had a quoted string with an
item in angle brackets, then we would mis-parse it. For instance,
"Bob <1234>" <1234@example.org>
would be misparsed as having the URI "1234"
The fix for this is to use parsing functions from reqresp_parser.h
since they are heavily tested and are awesome.
(issue ASTERISK-18990)
........
Merged revisions 351284 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Tue, 17 Jan 2012 16:07:42 +0000 (16:07 +0000)]
Fix udptl issue with initial INVITE introduced by r351027
When an inital INVITE occurs that contains image media, a channel
is not yet associated with the SIP dialog. The file descriptor
associated with the udptl session needs to be set in
initialize_udptl or in sip_new to account for this scenario.
........
Merged revisions 351233 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Russell Bryant [Tue, 17 Jan 2012 01:43:19 +0000 (01:43 +0000)]
Merged revisions 351182 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r351182 | russell | 2012-01-16 20:37:03 -0500 (Mon, 16 Jan 2012) | 22 lines
Add some missing locking in chan_sip.
This patch adds some missing locking to the function
send_provisional_keepalive_full(). This function is called from the scheduler,
which is processed in the SIP monitor thread. The associated channel (or pbx)
thread will also be using the same sip_pvt and ast_channel so locking must be
used. The sip_pvt_lock_full() function is used to ensure proper locking order
in a safe manner.
In passing, document a suspected reference counting error in this function.
The "fix" is left commented out because when the "fix" is present, crashes
occur. My theory is that fixing it is exposing a reference counting error
elsewhere, but I don't know where. (Or my analysis of this being a problem
could have been completely wrong in the first place). Leave the comment in
the code for so that someone may investigate it again in the future.
Also add a bit of doxygen to transmit_provisional_response().
Terry Wilson [Mon, 16 Jan 2012 21:17:08 +0000 (21:17 +0000)]
Ensure ACK retransmit & hangup on non-200 response to INVITE
When handling a non-2xx final response on an INVITE transaction, we have to
keep the transaction around after we send an ACK in case we receive a
retransmission of the response so we can re-transmit the ACK, but also tear
down the ast_channel as soon as we transmit the ACK. Before this patch, we
could fail at both of these things. Calling sip_alreadygone/needdestroy
prevented us from keeping the transaction up and retransmitting the ACK, and
queueing CONGESTION was not sufficient to cause the channel to be torn down
when originating calls via the CLI, for example.
This patch queues a hangup with CONGESTION instead of just queueing CONGESTION
for these responses and removes the sip_alreadygone and sip_needdestroy calls
from handle_response_invite on non-2xx responses. It relies on the hangup
calling sip_scheddestroy.
For more information, see section 17.1.1.1 of RFC 3261.
Terry Wilson [Mon, 16 Jan 2012 20:13:55 +0000 (20:13 +0000)]
Don't prematurely stop SIP session timer
When Asterisk is the UAS (incoming call, endpoint is re-inviting) the SIP session timer expires after half the time the sip endpoint indicates in the Session-expires header in proc_session_timer(). The session timer was being stopped totally and being handled as an error case instead of running again until the second expiry. This patch treats the half-time expiry as a non-error case and continues the timer until the true expiry.
(closes issue ASTERISK-18996)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
Patches: session_timer_fix.diff by Terry Wilson (License #5357)
based on session_timer.patch by Thomas Arimont (License #5525)
........
Merged revisions 351080 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Mon, 16 Jan 2012 19:12:38 +0000 (19:12 +0000)]
Create and initialize udptl only when dialog negotiates for image media
Prior to this patch, the udptl struct was allocated and initialized when a
dialog was associated with a peer that supported T.38, when a new SIP
channel was allocated, or what an INVITE request was received. This resulted
in any dialog associated with a peer that supported T.38 having udptl support
assigned to it, including the UDP ports needed for communication. This
occurred even in non-INVITE dialogs that would never send image media.
This patch creates and initializes the udptl structure only when the SDP
for a dialog specifies that image media is supported, or when Asterisk
indicates through the appropriate control frame that a dialog is to support
T.38.
(closes issue ASTERISK-16698)
Reported by: under
Tested by: Stefan Schmidt
Patches: udptl_20120113.diff uploaded by mjordan (License #6283)
Walter Doekes [Sun, 15 Jan 2012 20:12:54 +0000 (20:12 +0000)]
Allow only one thread at a time to do asterisk cleanup/shutdown.
Add locking around the really-really-quit part of the core stop/restart
part. Previously more than one thread could be called to do cleanup,
causing atexit handlers to be run multiple times, in turn causing
segfaults.
(issue ASTERISK-18883)
Reviewed by: Terry Wilson
Review: https://reviewboard.asterisk.org/r/1662/
Review: https://reviewboard.asterisk.org/r/1658/
........
Merged revisions 350888 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Kevin P. Fleming [Sat, 14 Jan 2012 16:41:55 +0000 (16:41 +0000)]
Ensure that all AC_LANG_PROGRAM calls in the configure script are properly quoted.
Recent versions of autoconf (2.68 on my system) won't properly process the configure
script unless every call to AC_LANG_PROGRAM is m4-quoted. Many calls in the script
were, but many were not. This patch corrects the unquoted calls.
........
Merged revisions 350837 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Kinsey Moore [Fri, 13 Jan 2012 21:41:24 +0000 (21:41 +0000)]
Make sure asterisk builds on OpenBSD
OpenBSD defines SO_PEERCRED, but it returns a 'struct sockpeercred', not
'struct ucred', which causes compilation of main/asterisk.c to fail in
read_credentials(). This allows configure to check for sockpeercred and
asterisk to deal with it properly.
(closes issue ASTERISK-18929) Reported-by: Barry Miller Patch-by: Barry Miller
........
Merged revisions 350730 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Fri, 13 Jan 2012 16:59:02 +0000 (16:59 +0000)]
Realtime queues failed to load queue information without queue member table
Previously, realtime queues could be loaded without defining the queue member
table. This allowed for queue members to be dynamic, while the realtime
queue definitions could exist in some backing storage. Revision 342223 broke
this when it changed the return value for realtime_multientry to return NULL
when no results are returned. Previously, an empty ast_config object was
expected.
(closes issue ASTERISK-19170)
Reported by: Rene Mendoza
Tested by: Rene Mendoza
Patches:
rt_queue_member_patch.diff uploaded by Matt Jordan (license 6283)
........
Merged revisions 350552 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Fri, 13 Jan 2012 16:43:11 +0000 (16:43 +0000)]
Fix crash from bridge channel hangup race condition in ConfBridge
This patch addresses two issues in ConfBridge and the channel bridge layer:
1. It fixes a race condition wherein the bridge channel could be hung up
2. It removes the deadlock avoidance from the bridging layer and makes the
bridge_pvt an ao2 ref counted object
Patch by David Vossel (mjordan was merely the commit monkey)
(issue ASTERISK-18988)
(closes issue ASTERISK-18885)
Reported by: Dmitry Melekhov
Tested by: Matt Jordan
Patches: chan_bridge_cleanup_v.diff uploaded by David Vossel (license 5628)
(closes issue ASTERISK-19100)
Reported by: Matt Jordan
Tested by: Matt Jordan
Richard Mudgett [Wed, 11 Jan 2012 21:47:04 +0000 (21:47 +0000)]
Make FollowMe optionally update connected line information when the accepting endpoint is bridged.
Like Dial and Queue, FollowMe needs to deal with
AST_CONTROL_CONNECTED_LINE information so when the parties are initially
bridged, the connected line information will be correct.
* Added the 'I' option just like the app_dial and app_queue 'I' option.
* Made 'N' option ignored if the call is already answered.
Walter Doekes [Mon, 9 Jan 2012 19:34:33 +0000 (19:34 +0000)]
Fix shutdown handling of sqlite3 astdb.
If a db_sync was scheduled just before shutdown, the atexit code calling
db_sync would have no effect, causing the astdb commit thread to stay
alive. This caused the SIP/realtime_sipregs test to fail. (The fallback
kill would run the atexit code again and that would wreak havoc.) This
fixes that the atexit kill condition is picked up properly.
(closes issue ASTERISK-18883)
Reviewed by: Terry Wilson
Update contrib script live_ast to invoke Asterisk with valgrind and suppression file.
* Added valgrind_compare script to compare two valgrind log files for
differences.
(issue ASTERISK-17339)
Reported by: Tzafrir Cohen
Patches:
valgrind_compare (license #5035) script uploaded by Tzafrir Cohen
live_ast_valgrind.diff (license #5035) patch uploaded by Tzafrir Cohen
live_ast_valgrind_v2.diff (license #5185) patch uploaded by Paul Belanger
........
r350128 | rmudgett | 2012-01-09 12:54:56 -0600 (Mon, 09 Jan 2012) | 11 lines
live_ast: valgrind: run asterisk under valgrind
Adds a new sub-command, "valgrind" to live_ast. It runs asterisk under
valgrind. The extra command-line parameters are passed to Asterisk as
usual, and parameters to valgrind are passed through LIVE_AST_VALGRIND_ARGS
in live.conf .
Review: https://reviewboard.asterisk.org/r/1109/
Merged revisions 326636 from http://svn.asterisk.org/svn/asterisk/branches/10
........
Merged revisions 350127-350128 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Kinsey Moore [Mon, 9 Jan 2012 15:39:31 +0000 (15:39 +0000)]
Prevent SLA settings from getting wiped out on reload
If SLA was reloaded without the config file being changed, current settings got
wiped out before the SLA reload code decided it wasn't going to reload the file
since nothing was changed. Moving the settings reset later in the reload
process fixes this.
(closes issue AST-744)
........
Merged revisions 350023 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Terry Wilson [Fri, 6 Jan 2012 23:25:03 +0000 (23:25 +0000)]
Don't leak CID in From header when presentation=unavailable
When someone does Set(CALLERPRES()=unavailable) (or
Set(CALLERID(pres)=unavailable)) when sendrpid=no, the From header shows
"Anonymous" <anonymous@anonymous.invalid>. When sendrpid=yes/pai, the From
header will still display the callerid info, even though we supply an rpid
header with the anonymous info. It seems like we shouldn't leak that info in
any case. Skimming http://tools.ietf.org/html/draft-ietf-sip-privacy-04 seems
to indicate that one shouldn't send identifying info in the From in this case.
This patch anonymizes the From header as well even when sendrpid=yes/pai.
Kinsey Moore [Fri, 6 Jan 2012 21:25:19 +0000 (21:25 +0000)]
Fix lua goto detection to prevent unexpected behavior with confbridge
A bug in the pbx_lua goto detection was causing the dialplan to hangup
unexpectedly after confbridge exited if it had called lua dialplan code during
execution.
Patch-by: Timo Teras Acked-by: Matt Nicholson
(closes issue ASTERISK-18976)
Matthew Jordan [Thu, 5 Jan 2012 23:56:52 +0000 (23:56 +0000)]
Fix premature free'ing of the frame committed in r349608
Even though we set the frame to the ast_null_frame and return that,
the caller of the frame hook may still need the frame. This now is
a bit more careful about when it frees the frame, i.e., only under
the same conditions that applied when we duplicated it in the first
place.
Richard Mudgett [Thu, 5 Jan 2012 23:46:01 +0000 (23:46 +0000)]
Make not assume that the cel_sqlite3_custom SQL table primary key is AcctId.
If a table is created by some other application and the primary key is not
named "AcctId", cel/cel_sqlite3_custom.c will always try to create the
table and fail because it already exists.
* Change the SQL table query to not require AcctId as the primary key.
Kinsey Moore [Thu, 5 Jan 2012 22:10:45 +0000 (22:10 +0000)]
Allow playback of formats that don't support seeking
ast_streamfile previously did unconditional seeking on files that broke
playback of formats that don't support that functionality. This patch avoids
the seek that was causing the problem. This regression was introduced in
r158062.
(closes issue ASTERISK-18994) Patch-by: Timo Teras
........
Merged revisions 349731 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Jonathan Rose [Thu, 5 Jan 2012 21:55:01 +0000 (21:55 +0000)]
Fix an issue where dsp.c would interpret multiple dtmf events from a single key press.
When receiving calls from a mobile phone into a DISA system on a connection with
significant interference, the reporter's Asterisk system would interpret DTMF incorrectly
and replicate digits received. This patch resolves that by increasing the number of
frames a mismatch has to be detected before assuming the DTMF is over by 1 frame and
adjusts dtmf_detect function to reset hits and misses only when an edge is detected.
(closes issue ASTERISK-17493)
Reported by: Alec Davis
Patches:
bug18904-refactor.diff.txt uploaded by Alec Davis (license 5546)
Review: https://reviewboard.asterisk.org/r/1130/
........
Merged revisions 349728 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Jonathan Rose [Thu, 5 Jan 2012 16:07:05 +0000 (16:07 +0000)]
Ensures Asterisk closes when receiving terminal signals in 'no fork' mode.
When catching a signal, in no fork mode the console thread is identical to the thread
responsible for catching the signal and closing Asterisk, which requires it to first
dispense with the console thread. Prior to this patch, if these threads were identical,
upon receiving a killing signal, the thread will send an URG signal to itself, which
we also catch and then promptly do nothing with. Obviously this isn't useful behavior.
(closes issue ASTERISK-19127)
Reported By: Bryon Clark
Patches:
quit_on_signals.patch uploaded by Bryon Clark (license 6157)
........
Merged revisions 349672 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Wed, 4 Jan 2012 22:19:34 +0000 (22:19 +0000)]
Fix for ConfBridge config parser unlocking channel mutex too many times
When looking up a ConfBridge profile, the config parser would, if it
found a channel datastore on the channel requesting the bridge profile,
unlock the channel mutex twice. Since that's a little aggressive,
it now only unlocks it once.
(closes issue ASTERISK-19042)
Reported by: Matt Jordan
Tested by: Matt Jordan
Patches:
19042 uploaded by David Vossel (license 5628)
Matthew Jordan [Wed, 4 Jan 2012 21:39:59 +0000 (21:39 +0000)]
Free successfully translated frame in fax_gateway_framehook
A frame that is translated via ast_translate is also duplicated via ast_frdup.
This will allocate a new frame on the heap, which needs to be free'd
at the appropriate time. This issue reporter used valgrind to find that this
occurred in res_fax's fax_gateway_framehook; a quick search through the code
showed that only place this was currently not handling the translatted frame
properly.
Richard Mudgett [Wed, 4 Jan 2012 20:50:24 +0000 (20:50 +0000)]
Fix segfault in chan_dahdi for CHANNEL(dahdi_span) evaluation on hangup.
* Added NULL private pointer checks in the following chan_dahdi channel
callbacks: dahdi_func_read(), dahdi_func_write(), dahdi_setoption(), and
dahdi_queryoption().
Kinsey Moore [Wed, 4 Jan 2012 20:23:45 +0000 (20:23 +0000)]
Make debian init script conform to the LSB standard
Previously, this init script would return 1 if Asterisk was already running.
This is incorrect behavior according to the LSB standard and has been fixed by
returning 0 instead.
Kinsey Moore [Wed, 4 Jan 2012 20:01:27 +0000 (20:01 +0000)]
Update autosupport script and man page
Added information collection from the output of the utilities: top, free, uptime, ifconfig
Added information collection from the output of the Asterisk command 'dahdi show status'
Added option / flag '-n, --non-interactive'
Updated man page to reflect new option / flag '-n, --non-interactive'
Patch-by: John Bigelow (itzanger)
(closes issue AST-749)
........
Merged revisions 349504 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Thu, 29 Dec 2011 15:14:08 +0000 (15:14 +0000)]
Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop
Failing to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop
causes the loop to exit prematurely. This causes a variety of negative side
effects, depending on when the loop exits. This patch handles the frame by
essentially swallowing the frame in the local loop, as the current channel
drivers expect the RTP bridge to handle the frame, and, in the case of the
local bridge loop, no additional action is necessary.
(issue ASTERISK-19040)
(issue ASTERISK-19128)
(issue ASTERISK-17725)
(issue ASTERISK-18340)
(closes issue ASTERISK-19095)
Reported by: Stefan Schmidt
Tested by: Matt Jordan
Kevin P. Fleming [Wed, 28 Dec 2011 18:49:29 +0000 (18:49 +0000)]
Improve T.38 gateway V.21 preamble detection.
This commit removes the V.21 preamble detection code previously added to the
generic DSP implementation in Asterisk, and instead enhances the res_fax module
to be able to utilize V.21 preamble detection functionality made available by
FAX technology modules. This commit also adds such support to res_fax_spandsp,
which uses the Spandsp modem tone detection code to do the V.21 preamble
detection.
There should be no functional change here, other than much more reliable V.21
preamble detection (and thus T.38 gateway initiation).
Matthew Jordan [Tue, 27 Dec 2011 20:53:46 +0000 (20:53 +0000)]
Fix timing source dependency issues with MOH
Prior to this patch, res_musiconhold existed at the same module priority level
as the timing sources that it depends on. This would cause a problem when
music on hold was reloaded, as the timing source could be changed after
res_musiconhold was processed. This patch adds a new module priority level,
AST_MODPRI_TIMING, that the various timing modules are now loaded at. This
now occurs before loading other resource modules, such that the timing source
is guaranteed to be set prior to resolving the timing source dependencies.
(closes issue ASTERISK-17474)
Reporter: Luke H
Tested by: Luke H, Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont
Patches:
asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-1.8.diff uploaded by elguero (License #5026)
asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-10.diff uploaded by elguero (License #5026)
asterisk-17474-dahdi_timing-infinite-wait-fix_v3.diff uploaded by elguero (License #5026)
Sean Bright [Tue, 27 Dec 2011 17:17:13 +0000 (17:17 +0000)]
Once an audiohook is attached to a channel, we continue to transcode all of the
frames, even after all of the hooks are detached. This patch short-cicuits us
out before we transcode unnecessarily.
........
Merged revisions 349144 from http://svn.asterisk.org/svn/asterisk/branches/1.8
In ChanSpy, don't create audiohooks that will never be used.
When ChanSpy is initialized it creates and attaches 3 audiohooks:
1) Read audio off of the channel that we are spying on
2) Write audio to the channel that we are spying on
3) Write audio to the channel that is bridged to the channel that we are
spying on.
The first is always necessary, but the others are used only when specific
options are passed to the ChanSpy application (B, d, w, and W to be specific).
When those flags are not passed, neither of those audiohooks are ever sent
frames, but we still try to process the hooks for each voice frame that we
recieve on the channel.
So in short - only create and attach audiohooks that we actually need.
........
projects / thirdparty / asterisk.git / log
