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.
........
Richard Mudgett [Fri, 23 Dec 2011 02:30:19 +0000 (02:30 +0000)]
Fix extension state callback references in chan_sip.
Chan_sip gives a dialog reference to the extension state callback and
assumes that when ast_extension_state_del() returns, the callback cannot
happen anymore. Chan_sip then reduces the dialog reference count
associated with the callback. Recent changes (ASTERISK-17760) have
resulted in the potential for the callback to happen after
ast_extension_state_del() has returned. For chan_sip, this could be very
bad because the dialog pointer could have already been destroyed.
* Added ast_extension_state_add_destroy() so chan_sip can account for the
sip_pvt reference given to the extension state callback when the extension
state callback is deleted.
* Fix pbx.c awkward statecbs handling in ast_extension_state_add_destroy()
and handle_statechange() now that the struct ast_state_cb has a destructor
to call.
* Ensure that ast_extension_state_add_destroy() will never return -1 or 0
for a successful registration.
* Fixed pbx.c statecbs_cmp() to compare the correct information. The
passed in value to compare is a change_cb function pointer not an object
pointer.
* Make pbx.c ast_merge_contexts_and_delete() not perform callbacks with
AST_EXTENSION_REMOVED with locks held. Chan_sip is notorious for
deadlocking when those locks are held during the callback.
* Removed unused lock declaration for the pbx.c store_hints list.
Matthew Jordan [Thu, 22 Dec 2011 22:37:31 +0000 (22:37 +0000)]
Fix for memory leaks / cleanup in cel_pgsql
There were a number of issues in cel_pgsql's pgsql_log method:
* If either sql or sql2 could not be allocated, the method would return while
the pgsql_lock was still locked
* If the execution of the log statement succeeded, the sql and sql2 structs
were never free'd
* Reconnection successes were logged as ERRORs. In general, the severity of
several logging statements was reduced
(closes issue ASTERISK-18879)
Reported by: Niolas Bouliane
Tested by: Matt Jordan
Terry Wilson [Thu, 22 Dec 2011 20:17:39 +0000 (20:17 +0000)]
Allow packetization vaules > 127
According to the RTP packetization documentation, and the maximum values
listed in AST_FORMAT_LIST, we should support values > that the signed
char array that ast_codec_pref makes available to store the value. All
places in the code treat the framing field as though it were an int
array instaead of a char array anyway, so this just fixes the type of
the array.
Richard Mudgett [Mon, 19 Dec 2011 21:37:31 +0000 (21:37 +0000)]
Fix crashes on other platforms caused by interference from Darwin weak symbol support.
Support weak symbols on a platform specific basis. The Mac OS X (Darwin)
support must be isolated from the other platforms because it has caused
other platforms to crash. Several other platforms including Linux have
GCC versions that define the weak attribute. However, this attribute is
only setup for use in the code by Darwin.
(closes issue ASTERISK-18728)
Reported by: Ben Klang
Richard Mudgett [Fri, 16 Dec 2011 23:56:50 +0000 (23:56 +0000)]
Clean-up on isle five for __ast_request_and_dial() and ast_call_forward().
* Add locking when a channel inherits variables and datastores in
__ast_request_and_dial() and ast_call_forward(). Note: The involved
channels are not active so there was minimal potential for problems.
* Remove calls to ast_set_callerid() in __ast_request_and_dial() and
ast_call_forward() because the set information is for the wrong direction.
* Don't use C++ keywords for variable names in ast_call_forward().
* Run the redirecting interception macro if defined when forwarding a call
in ast_call_forward(). Note: Currently will never execute because the
only callers that supply a calling channel supply a hungup or zombie
channel.
* Make feature_request_and_dial() put the transferee into autoservice when
it calls ast_call_forward() in case a redirection interception macro is
run. Note: Currently will never happen because the caller channel (Party
B) is always hungup at this time.
* Make feature_request_and_dial() ignore the AST_CONTROL_PROCEEDING frame
to silence a log message.
........
Merged revisions 348464 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Richard Mudgett [Fri, 16 Dec 2011 21:04:01 +0000 (21:04 +0000)]
Fix crash during CDR update.
The ast_cdr_setcid() and ast_cdr_update() were shown in ASTERISK-18836 to
be called by different threads for the same channel. The channel driver
thread and the PBX thread running dialplan.
* Add lock protection around CDR API calls that access an ast_channel
pointer.
Matthew Jordan [Wed, 14 Dec 2011 22:34:51 +0000 (22:34 +0000)]
Added support for all slin formats to app_originate
Previously, app_originate could not originate a call into a non-8kHz conference
bridge as the formats for non-8kHz slin codecs were not applied to the created
channel. This patch adds all of the formats by default, such that if a created
channel has a codec that supports a higher sampling rate, a translation path
can be built between it and other channels.
Matthew Jordan [Wed, 14 Dec 2011 21:58:12 +0000 (21:58 +0000)]
Fixed Asterisk crash when function QUEUE_MEMBER receives invalid input
The function QUEUE_MEMBER has two required parameters (queuename, option). It
was only checking for the presence of queuename. The patch checks for the
existence of the option parameter and provides better error logging when
invalid values are provided for the option parameter as well.
Richard Mudgett [Tue, 13 Dec 2011 23:06:11 +0000 (23:06 +0000)]
Fix FollowMe CallerID on outgoing calls.
The addition of the Connected Line support changed how CallerID is passed
to outgoing calls. The FollowMe application was not updated to pass
CallerID to the outgoing calls.
* Fix FollowMe CallerID on outgoing calls.
* Restructured findmeexec() to fix several memory leaks and eliminate some
duplicated code.
* Made check the return value of create_followme_number(). Putting a NULL
into the numbers list is bad if create_followme_number() fails.
* Fixed a couple uses of ast_strdupa() inside loops.
* The changes to bridge_builtin_features.c fix a similar CallerID issue
with the bridging API attended and blind transfers. (Not used at this
time.)
Stefan Schmidt [Tue, 13 Dec 2011 15:20:33 +0000 (15:20 +0000)]
Fix possible misshandling of an incoming SIP response as a peer poke response.
Also make sure peer has even qualify enabled when handle a peer poke response.
Terry Wilson [Mon, 12 Dec 2011 19:24:06 +0000 (19:24 +0000)]
Add a separate buffer for SRTCP packets
The function ast_srtp_protect used a common buffer for both SRTP and SRTCP
packets. Since this function can be called from multiple threads for the same
SRTP session (scheduler for SRTCP and channel for SRTP) it was possible for the
packets to become corrupted as the buffer was used by both threads
simultaneously.
This patch adds a separate buffer for SRTCP packets to avoid the problem.
(closes issue ASTERISK-18889, Reported/patch by Daniel Collins)
........
Merged revisions 347995 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Richard Mudgett [Fri, 9 Dec 2011 01:29:59 +0000 (01:29 +0000)]
Fix some parsing issues in add_exten_to_pattern_tree().
* Simplify compare_char() and avoid potential sign extension issue.
* Fix infinite loop in add_exten_to_pattern_tree() handling of character
set escape handling.
* Added buffer overflow checks in add_exten_to_pattern_tree() character
set collection.
* Made ignore empty character sets.
* Added escape character handling to end-of-range character in character
sets. This has a slight change in behavior if the end-of-range character
is an escape character. You must now escape it.
* Fix potential sign extension issue when expanding character set ranges.
* Made remove duplicated characters from character sets. The duplicate
characters lower extension matching priority and prevent duplicate
extension detection.
* Fix escape character handling when the escape character is trying to
escape the end-of-string. We could have continued processing characters
after the end of the exten string. We could have added the previous
character to the pattern matching tree incorrectly.
Walter Doekes [Thu, 8 Dec 2011 21:31:00 +0000 (21:31 +0000)]
Fix regression when using tcpenable=no and tlsenable=yes.
The tlsenable settings are tucked away in main/tcptls.c, so I missed
them when resolving ASTERISK-18837. This should resolve the test suite
breakage of the sip tls tests.
Review: https://reviewboard.asterisk.org/r/1615
Reviewed by: Matt Jordan
........
Merged revisions 347718 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Jonathan Rose [Thu, 8 Dec 2011 20:43:20 +0000 (20:43 +0000)]
Fix regressed behavior of queue set penalty to work without specifying 'in <queuename>'
r325483 caused a regression in Asterisk 10+ that would make Asterisk segfault when
attempting to set penalty on an interface without specifying a queue in the queue set
penalty CLI command. In addition, no attempt would be made whatsoever to perform the
penalty setting on all the queues in the core list with either the cli command or the
non-segfaulting ami equivalent. This patch fixes that and also makes an attempt to
document and rename some functions required by this command to better represent what
they actually do. Oh yeah, and the use of this command without specifying a specific
queue actually works now.
Richard Mudgett [Thu, 8 Dec 2011 17:53:24 +0000 (17:53 +0000)]
Mark channel running the h exten with the soft-hangup flag.
When a bridge is broken, ast_bridge_call() might execute the h exten on
the calling channel. However, that channel may not have been the channel
that broke the bridge by hanging up. The channel executing the h exten
must be in a hung up state so things like AGI run in the correct mode.
* Make sure ast_bridge_call() marks the channel it is executing the h
exten on as hung up. (The AST_SOFTHANGUP_APPUNLOAD flag is used so as to
match the pbx.c main dialplan execution loop when it executes the h
exten.)
(closes issue ASTERISK-18811)
Reported by: David Hajek
Patches:
jira_asterisk_18811_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: David Hajek, rmudgett
........
Merged revisions 347595 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Terry Wilson [Thu, 8 Dec 2011 16:20:25 +0000 (16:20 +0000)]
Don't crash on INFO automon request with no channel
AST-2011-014. When automon was enabled in features.conf, it was possible
to crash Asterisk by sending an INFO request if no channel had been
created yet.
(closes issue ASTERISK-18805)
........
Merged revisions 347530 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2
........
Merged revisions 347531 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Jonathan Rose [Wed, 7 Dec 2011 20:27:59 +0000 (20:27 +0000)]
Fix: Meetme recording variables from realtime DB use null entries over channel variables
Meetme would attempt to substitute the realtime values of RECORDING_FILE and
RECORDING_FORMAT from the meetme db entry instead of using the channel variable set
for those variables in spite of those database entries being NULL or even lacking
a column to represent them.
(closes issue ASTERISK-18873)
Reported by: Byron Clark
Patches:
ASTERISK-18873-1.patch uploaded by Byron Clark (license 6157)
........
Merged revisions 347369 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Jonathan Rose [Tue, 6 Dec 2011 21:53:00 +0000 (21:53 +0000)]
Documents CHANNEL(musicclass) taking priority over m([x]) in waitExten
If waitExten specifies a music class to use with its music on hold option, it will use
CHANNEL(musicclass) instead if that channel variable has been set on the initiating
channel. This documents that behavior in the waitExten app so that this can be known
without checking the documentation of the code in function local_ast_moh_start.
(closes issue ASTERISK-18804)
........
Merged revisions 347239 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Walter Doekes [Tue, 6 Dec 2011 19:42:53 +0000 (19:42 +0000)]
Don't allow transport=tcp when tcpenable=no.
When tcpenable=no, sending to transport=tcp hosts was still allowed.
Resolving the source address wasn't possible and yielded the string
"(null)" in SIP messages. Fixed that and a couple of not-so-correct
log messages.
(closes issue ASTERISK-18837)
Reported by: Andreas Topp
Review: https://reviewboard.asterisk.org/r/1585
Reviewed by: Matt Jordan
........
Merged revisions 347166 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Walter Doekes [Tue, 6 Dec 2011 19:20:10 +0000 (19:20 +0000)]
Move setting of voicemail zonetag and locale up a bit.
The voicemail [general] zonetag and locale variables weren't loaded
until after the mailboxes were initialized. This caused the settings to
be unset for those mailboxes until a reload was performed.
(closes issue ASTERISK-18838)
Review: https://reviewboard.asterisk.org/r/1570
Reviewed by: Matt Jordan
........
Merged revisions 347111 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Matthew Jordan [Tue, 6 Dec 2011 17:24:51 +0000 (17:24 +0000)]
Fixed crash from orphaned MWI subscriptions in chan_sip
This patch resolves the issue where MWI subscriptions are orphaned
by subsequent SIP SUBSCRIBE messages. When a peer is removed, either
by pruning realtime SIP peers or by unloading / loading chan_sip, the
MWI subscriptions that were orphaned would still be on the event engine
list of valid subscriptions but have a pointer to a peer that no longer
was valid. When an MWI event would occur, this would cause a seg fault.
(closes issue ASTERISK-18663)
Reported by: Ross Beer
Tested by: Ross Beer, Matt Jordan
Patches:
blf_mwi_diff_12_06_11.txt uploaded by Matt Jordan (license 6283)
Richard Mudgett [Mon, 5 Dec 2011 17:42:36 +0000 (17:42 +0000)]
Restore call progress code for analog ports.
Extracting sig_analog from chan_dahdi lost call progress detection
functionality.
* Fix analog ports from considering a call answered immediately after
dialing has completed if the callprogress option is enabled.
(closes issue ASTERISK-18841)
Reported by: Richard Miller
Patches:
chan_dahdi.diff (license #5685) patch uploaded by Richard Miller (Modified by me)
sig_analog.c.diff (license #5685) patch uploaded by Richard Miller (Modified by me)
sig_analog.h.diff (license #5685) patch uploaded by Richard Miller
........
Merged revisions 347006 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Jonathan Rose [Mon, 5 Dec 2011 15:02:00 +0000 (15:02 +0000)]
Resolve duplicate label used in multiple priorities for the same extension.
Prior to this patch, if labels with the same name were used for different priorities in
the same extension, the new label would be accepted, but it would be unusable since
attempts to reach that label would just go to the first one. Now pbx.c detects this,
generates a warning in logs, and culls the label before adding it to the dialplan.
Kinsey Moore [Mon, 5 Dec 2011 14:46:25 +0000 (14:46 +0000)]
Fix chan_jingle/gtalk load regression introduced in r346087
Add missing symbol exports for ast_aji_client_destroy and ast_aji_buddy_destroy
for usage outside res_jabber. Testing of these changes focused on res_jabber
itself, so this problem was missed.
Reported-by: Michael Spiceland
........
Merged revisions 346951 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Walter Doekes [Sun, 4 Dec 2011 10:03:31 +0000 (10:03 +0000)]
For SIP REGISTER fix domain-only URIs and domain ACL bypass.
The code that allowed admins to create users with domain-only uri's had
stopped to work in 1.8 because of the reqresp parser rewrites. This is
fixed now: if you have a [mydomain.com] sip user, you can register with
useraddr sip:mydomain.com. Note that in that case -- if you're using
domain ACLs (a configured domain list) -- mydomain.com must be in the
allow list as well.
Reviewboard r1606 shows a list of registration combinations and which
SIP response codes are returned.
Review: https://reviewboard.asterisk.org/r/1533/
Reviewed by: Terry Wilson
Matthew Jordan [Fri, 2 Dec 2011 23:27:10 +0000 (23:27 +0000)]
Update SIP MESSAGE To parsing to correctly handle URI
The previous patch (r346040) incorrectly parsed the URI in the presence
of a port, e.g., user@hostname:port would fail as the port would be
double appended to the SIP message. This patch uses the parse_uri function
to correctly parse the URI into its username and hostname parts, and places
them in the correct fields in the sip_pvt structure.
Richard Mudgett [Thu, 1 Dec 2011 21:14:55 +0000 (21:14 +0000)]
Re-resolve the STUN address if a STUN poll fails for res_stun_monitor.
The STUN socket must remain open between polls or the external address
seen by the STUN server is likely to change. However, if the STUN request
poll fails then the STUN server address needs to be re-resolved and the
STUN socket needs to be closed and reopened.
* Re-resolve the STUN server address and create a new socket if the STUN
request poll fails.
* Fix ast_stun_request() return value consistency.
* Fix ast_stun_request() to check the received packet for expected message
type and transaction ID.
* Fix ast_stun_request() to read packets until timeout or an associated
response packet is found. The stun_purge_socket() hack is no longer
required.
* Reduce ast_stun_request() error messages to debug output.
* No longer pass in the destination address to ast_stun_request() if the
socket is already bound or connected to the destination.
Cleaning up chan_sip/tcptls file descriptor closing.
This patch attempts to eliminate various possible instances of undefined behavior caused
by invoking close/fclose in situations where fclose may have already been issued on a
tcptls_session_instance and/or closing file descriptors that don't have a valid index
for fd (-1). Thanks for more than a little help from wdoekes.
(closes issue ASTERISK-18700)
Reported by: Erik Wallin
David Vossel [Tue, 29 Nov 2011 00:00:11 +0000 (00:00 +0000)]
Fixes memory leak in message API.
The ast_msg_get_var function did not properly decrement
the ref count of the var it retrieves. The way this is
implemented is a bit tricky, as we must decrement the var and then
return the var's value. As long as the documentation for the
function is followed, this will not result in a dangling pointer as
the ast_msg structure owns its own reference to the var while it
exists in the var container.
Walter Doekes [Wed, 23 Nov 2011 20:45:35 +0000 (20:45 +0000)]
Minor cleanup in chan_sip get_msg_text() function.
In r116240, get_msg_text() got an extra parameter to fix the unwanted
addition of trailing newlines to SIP MESSAGE bodies. This caused all
linefeeds to be trimmed, which isn't right either. This is a stop-gap;
the right fix is to return the original SIP request body.
Review: https://reviewboard.asterisk.org/r/1586
Reviewed by: Matt Jordan
........
Merged revisions 346147 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Kinsey Moore [Wed, 23 Nov 2011 17:14:41 +0000 (17:14 +0000)]
Fix res_jabber resource leaks
This should fix almost all resource leaks in res_jabber that involve
ASTOBJ_CONTAINER_FIND and resolves an ambiguous situation where
ast_aji_get_client would sometimes bump an object's refcount and sometimes not.
Matthew Jordan [Wed, 23 Nov 2011 16:19:47 +0000 (16:19 +0000)]
Fixed SendMessage stripping extension from To: header in SIP MESSAGE
When using the MessageSend application to send a SIP MESSAGE to a non-peer,
chan_sip attempted to validate the hostname or IP Address. In the process,
it stripped off the extension and failed to add it back to the sip_pvt
structure before transmitting. This patch adds the full URI passed in
from the message core to the sip_pvt structure.
(closes issue ASTERISK-18903)
Reported by: Shaun Clark
Tested by: Matt Jordan
Terry Wilson [Wed, 23 Nov 2011 16:10:07 +0000 (16:10 +0000)]
Resume playing existing hold music for cached realtime MOH
As a result of the fix for ASTERISK-18039, realtime caching MOH no longer
properly resumes playing back a file between different holds in the same call.
This is because scanning for new files causes the existing file array to be
emptied and we were just comparing that the saved pointer to the filename
matched the pointer to the filename in a particular position in the array. An
easy fix is to save the filename instead of a pointer to it and then do a
strcmp instead of comparing the addresses.
Richard Mudgett [Tue, 22 Nov 2011 23:00:31 +0000 (23:00 +0000)]
Fix dnsmgr entries to ask for the same address family each time.
The dnsmgr refresh would always get the first address found regardless of
the original address family requested. So if you asked for only IPv4
addresses originally, you might get an IPv6 address on refresh.
* Saved the original address family requested by ast_dnsmgr_lookup() to be
used when the address is refreshed.
........
Merged revisions 345976 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Terry Wilson [Mon, 21 Nov 2011 21:07:22 +0000 (21:07 +0000)]
Default to nat=yes; warn when nat in general and peer differ
It is possible to enumerate SIP usernames when the general and user/peer
nat settings differ in whether to respond to the port a request is sent
from or the port listed for responses in the Via header. In 1.4 and 1.6.2,
this would mean if one setting was nat=yes or nat=route and the other was
either nat=no or nat=never. In 1.8 and 10, this would mean when one was
nat=force_rport and the other was nat=no.
In order to address this problem, it was decided to switch the default
behavior to nat=yes/force_rport as it is the most commonly used option
and to strongly discourage setting nat per-peer/user when at all possible.
For more discussion of the issue, please see:
http://lists.digium.com/pipermail/asterisk-dev/2011-November/052191.html
Tilghman Lesher [Fri, 18 Nov 2011 22:16:26 +0000 (22:16 +0000)]
Fix a change in behavior in 'database show' from 1.8.
In 1.8 and previous versions, one could use any fullword portion of the key
name, including the full key, to obtain the record. Until this patch, this
did not work for the full key.
Closes issue ASTERISK-18886
Patch: by tilghman
Review: by twilson (http://pastebin.com/7rtu6bpk) on #asterisk-dev
Jonathan Rose [Wed, 16 Nov 2011 14:49:43 +0000 (14:49 +0000)]
Guarantee messages go into the right folders with multiple recipients
Before, using the U flag in Voicemail with multiple recipients would put urgent messages
in the INBOX folder for all users past the first thanks to a bug with the message
copying function. This would also cause messages to fail to be sent if the INBOX
directory hadn't been created for that mailbox yet.
(closes issue ASTERISK-18245)
Reported by: Matt Jordan
(closes issue ASTERISK-18246)
Reported by: Matt Jordan
Richard Mudgett [Mon, 14 Nov 2011 21:59:28 +0000 (21:59 +0000)]
Restore SIP DTMF overlap dialing method.
The recent fix for ASTERISK-17288 to get RFC3578 SIP overlap support
working correctly removed a long standing ability to do overlap dialing
using DTMF in the early media phase of a call.
See ASTERISK-18702 it has a very good description of the issue.
I started with Pavel Troller's chan_sip.diff patch on issue
ASTERISK-18702.
* Added 'dtmf' enum value to sip.conf allowoverlap config option. The new
option value causes the Incomplte application to not send anything with
chan_sip so the caller can supply more digits via DTMF.
* Renames SIP_GET_DEST_PICKUP_EXTEN_FOUND to SIP_GET_DEST_EXTEN_MATCHMORE
since that is what it really means.
* Fixed get_destination() inconsistency with the pickup extension
matching.
* Fixed initialization of PAGE3 of global_flags in reload_config().
(closes issue ASTERISK-18702)
Reported by: Pavel Troller
is only valid when new_frames == 1. Otherwise we start reading into adjacent
variables declared on the stack. The read end discards what is read, so the
values don't matter but it's not a good idea to read past where we want even
though new_frames is almost always 1 and should never be large. This patch is
basically taken out of kpfleming's eventfd branch, as he mentioned that he
remembered fixing it there when I talked to him about this issue.
Jonathan Rose [Mon, 14 Nov 2011 16:12:37 +0000 (16:12 +0000)]
Moves voicemail setup password entry to the end of the setup process.
This change was made because forcegreeting and forcename settings in voicemail could be
circumvented by hanging up after entering a password, because the only way voicemail
currently observes whether a mailbox is new or not is by checking to see if the password
is the same as the mailbox number or not.
(closes issue ASTERISK-18282)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1581/
........
Merged revisions 345062 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Kinsey Moore [Mon, 14 Nov 2011 15:10:38 +0000 (15:10 +0000)]
Ensure that a null vmexten does not cause a segfault
When sip_send_mwi_to_peer was modified recently to avoid deadlocks, vmexten
was not expected to be null. This change handles that situation to avoid
a segfault.
........
Merged revisions 345063 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Terry Wilson [Sat, 12 Nov 2011 00:34:40 +0000 (00:34 +0000)]
Don't forget to rescan MOH files for cached realtime classes
Realtime MOH class caching was implemented because without it, you would build
a completely new MOH class and would start the music over at the beginning each
time hold was pressed in a conversation. Unfortunately, this broke re-scanning
for file changes for realtime MOH classes. This patch corrects that issue.
Walter Doekes [Fri, 11 Nov 2011 21:58:31 +0000 (21:58 +0000)]
Use __alignof__ instead of sizeof for stringfield length storage.
Kevin P Fleming suggested that r343157 should use __alignof__ instead
of sizeof. For most systems this won't be an issue, but better fix it
now while it's still fresh.
Matthew Jordan [Fri, 11 Nov 2011 21:50:07 +0000 (21:50 +0000)]
Video format was treated as audio when removed from the file playback scheduler
This patch fixes the format type check in ast_closestream and
filestream_destructor. Previously a comparison operator was used, but since
audio formats are no longer contiguous (and AST_FORMAT_AUDIO_MASK includes
formats that have a value greater than the video formats), a bitwise AND
operation is used instead. Duplicated code was also moved to filestream_close.
(closes issue ASTERISK-18682)
Reported by: Aldo Bedrij
Tested by: Matt Jordan
Walter Doekes [Fri, 11 Nov 2011 21:36:01 +0000 (21:36 +0000)]
Remove unneeded if(params) checks in reqresp_parser.
Nick Lewis added them in https://reviewboard.asterisk.org/r/549/diff/1-2/
for no apparent reason. There is no way that params could become NULL in
that piece of code, so I removed these excess checks again.
........
Merged revisions 344837 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Walter Doekes [Fri, 11 Nov 2011 21:29:05 +0000 (21:29 +0000)]
Fix bad quoting of multiline mxml opaque_data that caused invalid xml.
The opaque_data was added and enclosed in single quotes, assuming it
would be only a single line. The rest of the lines were appended after
the closing quote.
(closes issue ASTERISK-18852)
Reported by: peep_ on IRC
Kinsey Moore [Fri, 11 Nov 2011 20:14:30 +0000 (20:14 +0000)]
Fix regression introduced by SDP fixups
If capability is adjusted when switching to UDPTL during fax transmission, fax
teardown fails. Make sure capability is only touched if RTP is active. This
regression was introduced in R344385.
........
Merged revisions 344769 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Richard Mudgett [Fri, 11 Nov 2011 18:01:36 +0000 (18:01 +0000)]
Make CLI "core show channel" not hold the channel lock during console output.
Holding the channel lock while the CLI "core show channel" command is
executing can slow down the system. It could block the system if the
console output is halted or paused.
* Made capture the CLI "core show channel" output into a buffer to be
output after the channel is unlocked.
* Removed use of C++ keyword as a variable name. out renamed to obuf.
* Checked allocation of obuf for failure so will not crash.
Richard Mudgett [Thu, 10 Nov 2011 23:01:31 +0000 (23:01 +0000)]
Fix potential deadlock calling ast_call() with channel locks held.
Fixed app_queue.c:ring_entry() calling ast_call() with the channel locks
held. Chan_local attempts to do deadlock avoidance in its ast_call()
callback and could deadlock if a channel lock is already held.
........
Merged revisions 344539 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Richard Mudgett [Thu, 10 Nov 2011 22:36:47 +0000 (22:36 +0000)]
Make AMI event AgentCalled get CallerID/ConnectedLine info from the incoming channel.
It was strange that the AgentCalled AMI event would get most of its
information from the incoming channel but then get the CallerID
information from the outgoing channel. Before connected line support was
added, this information was always the same at this point.
David Vossel [Thu, 10 Nov 2011 21:54:42 +0000 (21:54 +0000)]
Fixes issue with ConfBridge participants hanging up during DTMF feature menu usage getting stuck in conference forever.
When a conference user enters the DTMF menu they are suspended from the
bridge while the channel is handed off to the DTMF feature code. If a
user entered this state and hungup, there existed a race condition where
the channel could not exit the conference because it was waiting on a
signal that would never arrive. This patch fixes that, because it would
stupid for me to talk about the problem and commit a patch for something else.
Kinsey Moore [Thu, 10 Nov 2011 21:14:52 +0000 (21:14 +0000)]
Fix another incorrect case with meetme's PIN logic and add documentation
This fixes an issue where a user of a dynamic conference was asked for a PIN
twice. This also adds documentation to assist in future modifications to the
piece of code responsible for PIN checking.
(closes issue AST-670)
........
Merged revisions 344439 from http://svn.asterisk.org/svn/asterisk/branches/1.8
Kinsey Moore [Thu, 10 Nov 2011 18:14:20 +0000 (18:14 +0000)]
Fix several bugs with SDP parsing and well-formedness of responses
Fix bug ASTERISK-16558 which dealt with the order of responses to incoming
streams defined by SDP.
Fix unreported bug where offering multiple same-type streams would cause
Asterisk to reply with an incorrect SDP response missing one or more streams
without a proper declination.
Fix bugs related to a single non-audio stream being offered with responses
requesting codecs that were not offered in the initial invite along with an
additional audio stream that was not in the initial invite.
Terry Wilson [Wed, 9 Nov 2011 20:07:34 +0000 (20:07 +0000)]
Don't treat a host:port string as a domain
The domain matching code prior to 1.8 used to manually remove the port
from the host:port string when determining if an incoming request
matched the list of domains. When switching to the new parsing
functions, the documentation implied that the "domain" was being
returned by these functions, when instead it was returning the
"hostport" as defined by RFC 3261. This led to confusion and resulted
in 1.8+ rejecting an incoming request from x.x.x.x:xxxxx when
domain=x.x.x.x was set in sip.conf.
This patch renames the "domain" variables in the parsing functions to
"hostport" to more accurately describe what it is that they are
returning and also properly truncates the resulting hostport strings
when dealing with domain matching.
Kinsey Moore [Wed, 9 Nov 2011 17:14:28 +0000 (17:14 +0000)]
Fix pin parameter behavior regression in MeetMe
The last time this code was touched (by me), a subtlety was missed based on the
difference between needing to check a pin's validity and the need to prompt
for a pin.
(closes issue ASTERISK-18488)
........
Merged revisions 344102 from http://svn.asterisk.org/svn/asterisk/branches/1.8
projects / thirdparty / asterisk.git / log
