Terry Wilson [Wed, 7 Mar 2012 15:07:04 +0000 (15:07 +0000)]
Add detection for ODBC WCHAR fields
Without detecting these types, cel_odbc blows up when the character
set for the table is utf8. This also wraps cdr_adaptive_odbc's use of
those types in the HAVE_ODBC_WCHAR #ifdef seen in other parts of the
code.
........
Merged revisions 358435 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Richard Mudgett [Mon, 5 Mar 2012 21:38:50 +0000 (21:38 +0000)]
Setup DSP when SS7 call is connected or early media is available.
Outgoing SS7 calls fail to detect incoming DTMF so any bridged channel
that requires out-of-band DTMF will not work.
* Added sig_ss7_open_media() calls at appropriate places in sig_ss7.c.
The new call converts conditionaled out unconverted code and shows that
the code really did something useful.
* Improved some chan_dahdi DTMF debug messages to help track DTMF
handling.
(closes issue ASTERISK-19312)
Reported by: Igor Nikolaev
........
Merged revisions 358260 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Jonathan Rose [Mon, 5 Mar 2012 18:58:40 +0000 (18:58 +0000)]
Eliminate double close of file descriptor in manager.c
The process_output function in manager.c attempted to call fclose and close immediately
afterwards. Since fclose implies close, this resulted in a potential double free on file
descriptors. This patch changes that behavior and also adds error checking to fclose and
close depending on which was deemed necessary. Also error messages. Thanks to Rosen
Iliev for pointing out the location of the problem.
Kinsey Moore [Mon, 5 Mar 2012 15:59:46 +0000 (15:59 +0000)]
Ensure Asterisk acknowledges ACKs to 4xx on Replaces errors
Asterisk was not setting pendinginvite in the upper half of
handle_request_invite such that the 4xx was retransmitted repeatedly even
though an ack was received for every retransmission.
(closes issue ASTERISK-19303)
Reported by: Jon Tsiros
Patches:
fix-19303.patch uploaded by Jeremiah Gowdy (license 6358)
........
Merged revisions 358115 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Terry Wilson [Fri, 2 Mar 2012 23:28:21 +0000 (23:28 +0000)]
Fix unused-but-set-variable warnings
All of these were pretty obviously unused. Some were unused because
the code that used them was #if 0'd. In those cases, I just commented
out the unused-but-set variables.
........
Merged revisions 358029 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Kinsey Moore [Fri, 2 Mar 2012 21:03:11 +0000 (21:03 +0000)]
Fix case-sensitivity for device-specific event subscriptions and CCSS
This change fixes case-sensitivity for device-specific subscriptions such that
the technology identifier is case-insensitive while the remainder of the device
string is still case-sensitive. This should also preserve the original case of
the device string as passed in to the event system. CCSS is the only feature
affected as it is the only consumer of device-specific event subscriptions.
The second part of this patch addresses similar case-sensitivity issues within
CCSS itself that prevented it from functioning correctly after the fix to the
events system.
This adds a unit test to verify that the event system works as expected.
Richard Mudgett [Fri, 2 Mar 2012 18:37:15 +0000 (18:37 +0000)]
Remove ISDN hold restriction for non-bridged calls.
The check if an ISDN call is bridged before it could be placed on hold is
not necessary and is overly restrictive. The check was originally done to
prevent problems with call transfers in case a user tried to transfer a
call connected to an application to another call connected to an
application. The ISDN transfer code has not required this restriction for
quite some time because ECT could transfer any two active calls to each
other.
* Remove ISDN hold restriction for calls connected to applications.
* Made ast_waitfordigit_full() ignore AST_CONTROL_HOLD and
AST_CONTROL_UNHOLD instead of generating a warning message.
Mark Michelson [Fri, 2 Mar 2012 01:05:23 +0000 (01:05 +0000)]
Fix race condition that can cause important control frames (such as a hangup) to be missed.
This takes two actions.
1. Move the reading of the alertpipe in __ast_read() to immediately before the
removal of frames from the readq. This means we won't do something silly like
read from the alertpipe, then ignore the fact that there's a frame to get from
the readq since channel's fdno is the AST_TIMING_FD.
2. When ast_settimeout() sets the rate to 0 and the timingfunc to NULL, if the
channel's fdno is the AST_TIMING_FD, then set the fdno to -1. This is because
if the rate is 0 and the timingfunc is NULL, it means that the channel's timing
fd is being invalidated, so any pending reads should not occur.
This may actually solve more issues than the referenced one below, but it's not
known at this time for sure.
(closes issue ASTERISK-19223)
reported by Frank-Michael Wittig
Kinsey Moore [Thu, 1 Mar 2012 14:18:49 +0000 (14:18 +0000)]
Prevent outbound SIP NOTIFY packets from displaying a port of 0
In the change from 1.6.2 to 1.8, ast_sockaddr was introduced which changed the
behavior of ast_find_ourip such that port number was wiped out. This caused
the port in internip (which is used for Contact and Call-ID on NOTIFYs) to be
0. This change causes ast_find_ourip to be port-preserving again.
(closes issue ASTERISK-19430)
........
Merged revisions 357665 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Walter Doekes [Wed, 29 Feb 2012 20:39:39 +0000 (20:39 +0000)]
Update stringfield documentation for removed second va_list in favor of va_copy.
In r320946, the second va_list that was passed to ast_string_field_build_va
and friends, was removed. This patch updates the documentation to reflect that.
Walter Doekes [Wed, 29 Feb 2012 19:43:02 +0000 (19:43 +0000)]
Fix copying of CDR(accountcode) to local channels.
In r203638, during the addition of the Channel Event Logging, in mid-2009, this
got broken in trunk and ended up in asterisk 1.8 and higher. This fixes so the
CDR(accountcode) from the calling channel is available to dialed channels again
as well as showing up properly in the CDR's.
(closes issue ASTERISK-19384)
Reported by: jamicque
Patches: accountcode.patch (License #6033) by jamicque
Review: https://reviewboard.asterisk.org/r/1775/
Reviewed by: Richard Mudgett
........
Merged revisions 357575 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Jonathan Rose [Tue, 28 Feb 2012 22:29:47 +0000 (22:29 +0000)]
Adding transport=udp to sample sip.conf - Also changes version of Asterisk 1.8 in UPGRADE
(issue ASTERISK-19352)
Reported by: jamicque
Patches:
asterisk-19352-transport-warning-message-v1.patch uploaded by Michael L. Young (license 5026)
........
Merged revisions 357490 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Jonathan Rose [Tue, 28 Feb 2012 21:52:06 +0000 (21:52 +0000)]
Add additional character type types to supported data types for cdr_adaptive_odbc
The reporter was uable to use varchar utf8_unicode_ci with cdr_adaptive_odbc, so
this patch adds those along with some other character types to the list of types
cdr_adaptive_odbc will work using the varchar conditions. The problem wasn't really
UTF8 characters as much as it was a failure to respond to the exact type that was
declared/in use on that database.
(closes issue ASTERISK-19334)
Reported By: Igor Nikolaev
Patches:
cdr_adaptive_odbc.patch uploaded by Igor Nikolaev (license 6236)
........
Merged revisions 357455 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Tilghman Lesher [Tue, 28 Feb 2012 21:21:14 +0000 (21:21 +0000)]
Correctly reset the dialplan priority.
When the stack frame is allocated, we save the address to which we should
return, when the Gosub returns. However, if we just want to restore the
priority, then we need to subtract 1 before setting it. Otherwise, when
a Gosub goes to a nonexistent address, it will skip a priority in the
dialplan. This is because when we return from an application, the PBX
increments the priority for us.
........
Merged revisions 357416 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Jonathan Rose [Tue, 28 Feb 2012 18:11:15 +0000 (18:11 +0000)]
Changes transport option in sip.conf so that using multiple instances doesn't stack.
Prior to this patch, Using "transport=" multiple times would cause them to add to one
another like allow/deny. This patch changes that behavior to simply use the transport
option specified last. Also, if no transport option is applied now, the default will
automatically be UDP.
(closes ASTERISK-19352)
Reported by: jamicque
Patches:
asterisk-19352-transport-warning-message-v1.patch uploaded by Michael L. Young (license 5026)
issueA19352_no_transport_is_udp.patch uploaded by Walter Doekes (license 5674)
Review: https://reviewboard.asterisk.org/r/1745/diff/#index_header
........
Merged revisions 357266 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Kevin P. Fleming [Tue, 28 Feb 2012 14:46:15 +0000 (14:46 +0000)]
Make COMPILE_DOUBLE magic actually work.
The build system has some special magic to ensure that if Asterisk is built
with --enable-dev-mode *and* DONT_OPTIMIZE, that all the source is still compiled
with the optimizer enabled (even though the result will be thrown away), because
the compiler is able to find a great deal of coding errors and bugs as a result
of running its optimizers. Unfortunately at some point this mode got broken,
and the 'throwaway' compile of the code was no longer done with the optimizer
enabled. This patch corrects that problem.
........
Merged revisions 357212 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Sat, 25 Feb 2012 17:22:14 +0000 (17:22 +0000)]
Fix crash in app_voicemail during close_mailbox
In r354890, a memory leak in app_voicemail was fixed by properly disposing of
the allocated heard/deleted pointers. However, there are situations,
particularly when no messages are found in a folder, where these pointers are
not allocated and not NULL. In that case, an invalid free would be attempted,
which could crash app_voicemail. As there are a number of code paths where
this could occur, this patch uses the number of messages detected in the folder
before it attempts to free the pointers. This resolves the crash detected in
the Asterisk Test Suite's check_voicemail_nominal test.
........
Merged revisions 356797 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Fri, 24 Feb 2012 17:42:53 +0000 (17:42 +0000)]
Remove srtp_shutdown from res_srtp
The patch for ASTERISK-19253 included properly shutting down the libsrtp
library in the case of module unload. Unfortunately, not all distributions
have the srtp_shutdown call. As such, this patch removes calling
srtp_shutdown.
........
Merged revisions 356650 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Fri, 24 Feb 2012 15:07:41 +0000 (15:07 +0000)]
Allow SRTP policies to be reloaded
Currently, when using res_srtp, once the SRTP policy has been added to the
current session the policy is locked into place. Any attempt to replace an
existing policy, which would be needed if the remote endpoint negotiated a new
cryptographic key, is instead rejected in res_srtp. This happens in particular
in transfer scenarios, where the endpoint that Asterisk is communicating with
changes but uses the same RTP session.
This patch modifies res_srtp to allow remote and local policies to be reloaded
in the underlying SRTP library. From the perspective of users of the SRTP API,
the only change is that the adding of remote and local policies are now added
in a single method call, whereas they previously were added separately. This
was changed to account for the differences in handling remote and local
policies in libsrtp.
Review: https://reviewboard.asterisk.org/r/1741/
(closes issue ASTERISK-19253)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
Patches:
srtp_renew_keys_2012_02_22.diff uploaded by Matt Jordan (license 6283)
(with some small modifications for this check-in)
........
Merged revisions 356604 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Richard Mudgett [Thu, 23 Feb 2012 19:52:39 +0000 (19:52 +0000)]
Fix blind transfer parking issues if the dialed extension is not recognized as a parking extension.
Custom parking extensions may not be coded such that the first and only
extension priority is the Park application. These custom parking
extensions will not be recognized as parking extensions. When a call is
blind transferred to an extension that is not recognized as a parking
extension, the normal blind transfer code causes the transferred channel
to start executing dialplan. Calls that get parked in this manner do not
know the original channel name that parked the call so the original parker
could never be called back if the parked call is not retrieved before the
timeout time. The parking space is also announced to the call being
parked as a side effect of not knowing the original parking channel.
* Fix handling of BLINDTRANSFER channel variable for call parking.
* Fixed SIP blind transfer using the wrong dialplan context variable to
check for the parking extension.
Mark Michelson [Thu, 23 Feb 2012 15:40:23 +0000 (15:40 +0000)]
Fix ACK routing for non-2xx responses.
When we send an ACK for a 2xx response to an INVITE, we are supposed
to use the learned route set. However, when we receive a non-2xx final
response to an INVITE, we are supposed to send the ACK to the same place
we initially sent the INVITE.
We had been doing this up until the changes went in that would build a route
set from provisional responses. That introduced a regression where we would
use the learned route set under all circumstances.
With this change, we now will set the destination of our ACK based on the
invitestate. If it is INV_COMPLETED then that means that we have received
a non-2xx final response (INV_TERMINATED indicates a 2xx response was received).
If it is INV_CANCELLED, then that means the call is being canceled, which
means that we should be ACKing a 487 response.
The other change introduced here is setting the invitestate to INV_CONFIRMED
when we send an ACK *after* the reqprep instead of before. This way, we can
tell in reqprep more easily what the invitestate is prior to sending the ACK.
(closes issue ASTERISK-19389)
reported by Karsten Wemheuer
patches:
ASTERISK-19389v2.patch uploaded by Mark Michelson (license #5049)
(with some slight modifications prior to commit)
........
Merged revisions 356475 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Terry Wilson [Wed, 22 Feb 2012 21:18:22 +0000 (21:18 +0000)]
Track module use count for res_calendar
If the res_calendar module was followed immediately by one of the
calendar tech modules and "core stop gracefully" was run, Asterisk
would crash.
This patch adds use count tracking for res_calendar so that it is
unloaded after the tech modules when shutting down gracefully. It
is now not possible to unload all the of the calendar modules via
"module unload res_calednar.so", but it is still possible to unload
them all via "module unload -h res_calendar.so".
Fix potential buffer overrun and memory leak when executing "sip show peers"
The "sip show peers" command uses a fix sized array to sort the current peers
in the peers ao2_container. The size of the array is based on the current
number of peers in the container. However, once the size of the array is
determined, the number of peers in the container can change, as the peers
container is not locked. This could cause a buffer overrun when populating
the array, if peers were added to the container after the array was created.
Additionally, a memory leak of the allocated array would occur if a user
caused the _show_peers method to return CLI_SHOWUSAGE.
We now create a snapshot of the current peers using an ao2_callback with the
OBJ_MULTIPLE flag. This size of the array is set to the number of peers
that the iterator will iterate over; hence, if peers are added or removed
from the peers container it will not affect the execution of the "sip show
peers" command.
Review: https://reviewboard.asterisk.org/r/1738/
(closes issue ASTERISK-19231)
(closes issue ASTERISK-19361)
Reported by: Thomas Arimont, Jamuel Starkey
Tested by: Thomas Arimont, Jamuel Starkey
Patches: sip_show_peers_2012_02_16.diff uploaded by mjordan (license 6283)
........
Sean Bright [Mon, 20 Feb 2012 18:39:22 +0000 (18:39 +0000)]
Remove spurious warning when 'qualifyfreqnotok' is set successfully.
(closes issue ASTERISK-17176)
Reported by: John Covert
Tested by: Sean Bright
Patches:
chan_iax2.c.qualifyfreqnotok.patch uploaded by John Covert (license 5512)
........
Merged revisions 355997 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Alec L Davis [Sat, 18 Feb 2012 07:58:43 +0000 (07:58 +0000)]
push 'outgoing' flag from sig_XXX up to chan_dahdi
'p->outgoing' in chan_dahdi and sig_analog wern't kept in sync, particulary FXS ast_hangup didn't clear the 'outgoing' flag.
sig_pri and sig_ss7 were keeping 'outgoing' flag insync.
Now provides a callback for all the low level sig_XXX modules.
(issue ASTERISK-19316)
alecdavis (license 585)
Reported by: Jeremy Pepper
Tested by: alecdavis
Sean Bright [Fri, 17 Feb 2012 19:34:17 +0000 (19:34 +0000)]
Pass the correct value to ast_timer_set_rate() for IAX2 trunking.
IAX2 uses the trunkfreq variable to determine how often to send trunk packets, but
this value is in milliseconds while ast_timer_set_rate() expects the rate argument
to be ticks per second. So we divide 1000 by trunkfreq and pass that in instead.
With a default of 20ms, this change makes IAX2 send trunk packets every 20ms
instead of every 50ms.
Tracked down by myself and Bob Wienholt.
........
Merged revisions 355746 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Mark Michelson [Fri, 17 Feb 2012 19:06:57 +0000 (19:06 +0000)]
Fix regressions with regards to route-set creation on early dialogs.
This fixes two main issues:
1. Asterisk would send a CANCEL to the route created by the provisional response
instead of using the same destination it did in the initial INVITE.
2. If a new route set arrives in a 200 OK than was in the 1XX response (perfectly
possible if our outbound INVITE gets forked), then the route set in the 200 OK
needs to overwrite the route set in the 1XX response.
(closes issue ASTERISK-19358)
Reported by: Karsten Wemheuer
Tested by: Karsten Wemheuer
patches:
ASTERISK-19358.patch uploaded by Mark Michelson (license 5049)
ASTERISK-19358.patch uploaded by Stefan Schmidt (license 6034)
Sean Bright [Thu, 16 Feb 2012 20:01:59 +0000 (20:01 +0000)]
Revert a change to audio_audiohook_write_list that had no affect.
When I made this change initially, I was under the false impression that the
audiohooks structure remained on the channel after all of the hooks had been
detached. This is not the case, ast ast_read takes care of removing the
audiohooks structure if the lists are empty.
........
Merged revisions 355622 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Richard Mudgett [Thu, 16 Feb 2012 19:44:44 +0000 (19:44 +0000)]
Fix compile problem when old version of libvorbisfile v1.1.2 is used.
The principle difference between libvorbisfile v1.1.2 and newer (at least
v1.2.0) is the addition of the predefined callbacks OV_CALLBACKS_xxx in
vorbis/vorbisfile.h used for ov_open_callbacks().
* Updated the configure script to detect if libvorbisfile.h declares
OV_CALLBACKS_NOCLOSE.
* Copied the declaration of OV_CALLBACKS_NOCLOSE from v1.2.0 to allow
v1.1.2 to compile.
(closes issue ASTERISK-19370)
Reported by: Jonn Taylor
........
Merged revisions 355608 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Sean Bright [Wed, 15 Feb 2012 17:25:40 +0000 (17:25 +0000)]
Use TRUNK_CALL_START as originally intended.
Back in r646, TRUNK_CALL_START was added and defined as 0x4000. That same value
was also hard-coded in one part of the IAX2 code instead of using the #define.
TRUNK_CALL_START has changed over the years (for dealing with LOW_MEMORY), but
the hard-coded usage was never updated to match. This patch fixes that.
........
Merged revisions 355448 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Sean Bright [Tue, 14 Feb 2012 13:33:51 +0000 (13:33 +0000)]
Clear the high order bit from the destination call number before sending.
send_apathetic_reply takes the incoming frame's source call number as the
destination call number for the outgoing frame. If the incoming frame was a
full frame, then the high order bit of the source call number is set and will be
interpreted as a retransmit when sent back out as the destination call number.
........
Merged revisions 355182 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Since the dir timestamp is available at one second resolution, we cannot
know if it was updated within the same second after we scanned it.
Therefore, we will force another scan if the dir was just modified.
* Changed to force another scan if the directory was just modified.
Richard Mudgett [Mon, 13 Feb 2012 17:24:01 +0000 (17:24 +0000)]
Fix reconnecting to pgsql database after connection loss.
There can only be one database connection in res_config_pgsql just like
res_config_sqlite. If the connection is lost, the connection may not get
reestablished to the same database if the res_pgsql.conf and
extconfig.conf files are inconsistent.
* Made only use the configured database from res_pgsql.conf.
* Fixed potential buffer overwrite of last[] in config_pgsql().
(closes issue ASTERISK-16982)
Reported by: german aracil boned
Jason Parker [Fri, 10 Feb 2012 22:00:10 +0000 (22:00 +0000)]
Fix a voicemail memory leak with heard/deleted messages.
open_mailbox() was changed quite a long time ago to allocate this memory.
close_mailbox() should have been changed to be responsible for freeing it.
........
Merged revisions 354889 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Terry Wilson [Thu, 9 Feb 2012 22:03:51 +0000 (22:03 +0000)]
Note that CDRs are immutable once a bridge is torn down
CDRs cannot be modified after a bridge is torn down, (e.g. after
Dial() returns) even though the CDR() function may be called. Since
modifying the CDR code to change this behavior could very easily
break all kinds of things, this patch just documents this limitation.
Kinsey Moore [Thu, 9 Feb 2012 20:51:34 +0000 (20:51 +0000)]
Fix parsing of SIP headers where compact and non-compact headers are mixed
Change parsing of SIP headers so that compactness of the header no longer
influences which header will be chosen. Previously, a non-compact header
would be chosen instead of a preceeding compact-form header.
Kinsey Moore [Thu, 9 Feb 2012 19:54:04 +0000 (19:54 +0000)]
Make the config parser remove escaping backslashes
The config parser in Asterisk does not currently remove a backslash that is
used to escape a semicolon which would otherwise be interpreted as the start
of a comment.
The change here causes that backslash to be removed, but does not create a
real escape system in the config parser. The biggest complication with a real
escape system would be breaking existing configs everywhere (parsing \\ as \
and breaking on escaped non-semicolon characters) even though it would be the
"right" way to do things.
Matthew Jordan [Thu, 9 Feb 2012 16:35:43 +0000 (16:35 +0000)]
Fix SIP INFO DTMF handling for non-numeric codes
In ASTERISK-18924, SIP INFO DTMF handlingw as changed to account for both
lowercase alphatbetic DTMF events, as well as uppercase alphabetic DTMF
events. When this occurred, the comparison of the character buffer containing
the event code was changed such that the buffer was first compared again '0'
and '9' to determine if it was numeric. Unfortunately, since the first
character in the buffer will typically be '1' in the case of non-numeric
event codes (10-16), this caused those codes to be converted to a DTMF event
of '1'. This patch fixes that, and cleans up handling of both
application/dtmf-relay and application/dtmf content types.
Russell Bryant [Thu, 9 Feb 2012 02:25:28 +0000 (02:25 +0000)]
Remove some unnecessary locking from ast_hangup().
This patch removes some unnecessary locking of the channels container in
ast_hangup(). The reason this came up is that this lock can very quickly block
the entire system. If any of the channel cleanup code decides to block, it
causes a problem for the whole system. For example, when audiohooks get
destroyed, if that blocks for a while waiting on the mixmonitor thread to exit
because it's busy blocking on some I/O, it causes a problem for many other
threads in the meantime.
Terry Wilson [Tue, 7 Feb 2012 21:17:10 +0000 (21:17 +0000)]
Fix multiple SIP realtime issues
1. Set lastms to 0 when clearing instead of ""
2. Don't set ipaddr or port to the string "(null)" when they are empty
3. Add missing required fields, set default for lastms to 0, and modify
the length of the ipaddr field to 45 in the Postgresql realtime.sql
file.
Jonathan Rose [Tue, 7 Feb 2012 15:19:51 +0000 (15:19 +0000)]
Fix column duplication bug in module reload for cdr_pgsql.
Prior to this patch, attempts to reload cdr_pgsql.so would cause the column list to keep
its current data and then add a second copy during the reload. This would cause attempts
to log the CDR to the database to fail. This patch also cleans up some unnecessary null
checks for ast_free and deals with a few potential locking problems.
(closes issue ASTERISK-19216)
Reported by: Jacek Konieczny
Review: https://reviewboard.asterisk.org/r/1711/
........
Merged revisions 354263 from http://svn.asterisk.org/svn/asterisk/branches/1.8
* Allow acceptance of command without the app-data value. There are many
applications that do no need any parameters so it is silly to require that
field for all commands.
* Fixed a couple ast_malloc/ast_free mismatches with ast_add_extension2()
calls.
Jonathan Rose [Fri, 3 Feb 2012 21:25:27 +0000 (21:25 +0000)]
Fixes deadlocks occuring in chan_agent due to r335976
Bad locking order was added to chan_agent to prevent segfaults from having no locking
in a patch by irroot. This patch addresses the bad locking order by releasing locks before
getting the right locking order to stop deadlocks from occuring when doing multiple
interactions with agents.
Kinsey Moore [Thu, 2 Feb 2012 22:27:42 +0000 (22:27 +0000)]
Ensure entering T.38 passthrough does not cause an infinite loop
After R340970 Asterisk was still polling the RTCP file descriptor after RTCP is
shut down and removed. If the descriptor happened to have data ready when the
removal occured then Asterisk would go into an infinite loop trying to read
data that it can never actually access. This change disables the audio RTCP
file descriptor for the duration of the T.38 transaction.
Mark Michelson [Thu, 2 Feb 2012 18:48:05 +0000 (18:48 +0000)]
Fix TLS port binding behavior as well as reload behavior:
* Removes references to tlsbindport from http.conf.sample and manager.conf.sample
* Properly bind to port specified in tlsbindaddr, using the default port if specified.
* On a reload, properly close socket if the service has been disabled.
A note has been added to UPGRADE.txt to indicate how ports must be set for TLS.
(closes issue ASTERISK-16959)
reported by Olaf Holthausen
(closes issue ASTERISK-19201)
reported by Chris Mylonas
(closes issue ASTERISK-19204)
reported by Chris Mylonas
Jonathan Rose [Thu, 2 Feb 2012 18:32:07 +0000 (18:32 +0000)]
Blocked revisions 353818
........
Backports some documentation for func_curl from 10 to 1.8
For some reason this function was completely undocumented in 1.8. I copied the
10 docs over to 1.8 and removed references to an enumerator that was added in
the Asterisk 10 version of func_curl. That was the only change I noted.
Jonathan Rose [Thu, 2 Feb 2012 17:06:41 +0000 (17:06 +0000)]
Fix sip show peers port output, align columns, and fix ami port output.
A previous patch I committed from ASTERISK-16930 unexpectedly changed some output for
the AMI action "sippeers" which this patch changes back. Also, this aligns the output
for the cli command "sip show peers" and fixes another issue that patch introduced by
using ast_sockaddr_stringify calls multiple times without immediately using the pointer.
I also went ahead and did a little janitorial work to clean up whitespace in
_sip_show_peers.
(issue ASTERISK-16930)
(closes issue ASTERISK-19281)
Reported by: Patrick El Youssef
Patches:
ASTERISK-19281.diff uploaded by Walter Doekes (license 5674)
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Merged revisions 353769 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Jonathan Rose [Wed, 1 Feb 2012 21:16:53 +0000 (21:16 +0000)]
Use ast_sockaddr_stringify_fmt wrappers for various functions in chan_sip
There are a number of cleaner looking wrappers for ast_sockaddr_stringify_fmt
available which are slightly more readable than using a direct call to
ast_sockaddr_stringify_fmt. This patch switches a number of those calls in
chan_sip to use those wrappers and is generally harmless.
(Closes issue ASTERISK-16930)
Reported by: Michael L. Young
Patches:
chan_sip-broken-registration-1.8.diff uploaded by Michael L. Young (license 5026)
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Merged revisions 353720 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Sean Bright [Wed, 1 Feb 2012 15:51:29 +0000 (15:51 +0000)]
Resolve an overlap in the ast_audiohook_flags values.
AST_AUDIOHOOK_TRIGGER_WRITE and AST_AUDIOHOOK_WANTS_DTMF were overlapping which
may have caused unintended side effects. This patch moves
AST_AUDIOHOOK_TRIGGER_WRITE, and updates AST_AUDIOHOOK_TRIGGER_MODE to reflect
the original intention.
This will affect existing modules that use these flags, so be sure to recompile
as necessary.
Matthew Jordan [Wed, 1 Feb 2012 15:05:34 +0000 (15:05 +0000)]
Added clarification for the VERBOSITY setting to etc_default_asterisk
Clarified that using the VERBOSITY setting in etc_default_asterisk is the
same as using the -v command line switch, which causes Asterisk to launch
in console mode.
(closes issue ASTERISK-17030)
Reported by: Jonas
........
Merged revisions 353550 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Terry Wilson [Wed, 1 Feb 2012 00:00:02 +0000 (00:00 +0000)]
Allow res_calendar to be unloaded
The calendaring tech modules depend on res_calendar and initially
res_calendar just bumped the use count so that it couldn't be unloaded.
res_calendar can potentially create many threads and I've seen issues
where the Asterisk shutdown has failed where it looked like these
threads could be the culprit.
This patch adds unload support for res_calendar. Unloading res_calendar
will also unload the dependant tech modules as well.
projects / thirdparty / asterisk.git / log
