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.
(closes issue ASTERISK-18844)
Reported by: rmudgett
Review: https://reviewboard.asterisk.org/r/1635/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348940 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
Review: https://reviewboard.asterisk.org/r/1624/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348888 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
(closes issue ASTERISK-18876)
Review: https://reviewboard.asterisk.org/r/1639/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348833 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Some ISDN switches complain or block the call if the RDNIS number is
empty.
* Made chan_iax2 not save a RDNIS number into the ast_channel if the
string is blank. This is what other channel drivers do.
(closes issue ASTERISK-17152)
Reported by: rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348735 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
Review: https://reviewboard.asterisk.org/r/1617/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348647 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348464 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
(closes issue ASTERISK-18836)
Reported by: gpluser
Review: https://reviewboard.asterisk.org/r/1628/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348362 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ParkAndAnnounce tried to pass the CallerID to the announcing channel but
the ID was wiped out by the channel masquerade done when parking the call.
* Save the CallerID before parking the channel to pass it to the
announcing channel.
* Fixed a minor memory leak in ParkAndAnnounce.
* Updated some ParkAndAnnounce log messages.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348310 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.)
(closes issue ASTERISK-17557)
Reported by: hamlet505a
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1612/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348101 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@347995 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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.
(closes issue ASTERISK-18909)
Reported by: Luke-Jr
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@347811 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@347595 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@347369 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@347239 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@347166 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@347111 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
Review: https://reviewboard.asterisk.org/r/1610/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@347058 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@347006 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
(closes issue ASTERISK-18807)
Reported by: Kenneth Shumard
Patches:
pbx.c.patch uploaded by Kenneth Shumard (License 5077)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@346954 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@346951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
(closes issue ASTERISK-18389)
(closes issue ASTERISK-18741)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@346899 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
(closes issue ASTERISK-18327)
Reported by: Wolfram Joost
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1595/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@346700 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
(issue ASTERISK-18345)
Reported by: Stephane Cazelas
(issue ASTERISK-18342)
Reported by: Stephane Chazelas
Review: https://reviewboard.asterisk.org/r/1576/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@346564 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Update the documentation surrounding the use of MONITOR_EXEC to make it more clear
that it can be used for both Monitor() and MixMonitor() usage.
(closes issue ASTERISK-17413)
Reported by: David Woolley
Patches:
issue18817_mixmonitor_queues_doc.diff by Michael L. Young (License #5026)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@346472 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@346147 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
(closes issue ASTERISK-18912)
Review: https://reviewboard.asterisk.org/r/1596/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@346030 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@345976 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
(closes issue ASTERISK-18862)
Review: https://reviewboard.asterisk.org/r/1591/
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Merged revisions 345776 from http://svn.asterisk.org/svn/asterisk/branches/1.4
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Merged revisions 345800 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@345828 65c4cc65-6c06-0410-ace0-fbb531ad65f3
