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.
(issue ASTERISK-19192)
Review: https://reviewboard.asterisk.org/r/1681/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352014 65c4cc65-6c06-0410-ace0-fbb531ad65f3
gcc version 4.6.2 caught an unused variable in the ilbc codec
library. This would prevent compilation with --enable-dev-mode;
variable removed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@351760 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@351618 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@351611 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
(issue ASTERISK-19057)
Reported by: Yuri
Review: https://reviewboard.asterisk.org/r/1678/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@351559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
Review: https://reviewboard.asterisk.org/r/1663/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@351287 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@351284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@351233 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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().
(closes issue ASTERISK-18979)
Review: https://reviewboard.asterisk.org/r/1648
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@351182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
(closes issue ASTERISK-17717)
Review: https://reviewboard.asterisk.org/r/1672/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@351130 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@351080 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
(closes issue ASTERISK-16794)
Reported by: Elazar Broad
Tested by: Stefan Schmidt
review: https://reviewboard.asterisk.org/r/1668/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@351027 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Note that I'm not confirming legitimacy of having that file in tree at
all. Is anyone using aelparse/conf2ael?
(issue ASTERISK-15350)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350885 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350837 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Don't specify a specific version of libgmime; newer versions are available
now and acceptable.
* Install libsrtp so that res_srtp can be built.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350788 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350730 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Change eventextra to extra in cel_psql.c and cel_odbc.c.
* Change EventExtra to Extra in cel_manager.c.
(issue ASTERISK-17190)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350571 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Add missing eventextra to cel_psql.c and cel_odbc.c.
* Add missing PeerAccount and EventExtra to cel_manager.c.
* Add missing userdeftype support for cel_custom.conf.sample and
cel_sqlite3_custom.conf.sample.
(closes issue ASTERISK-17190)
Reported by: Bryant Zimmerman
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350555 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350552 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
(closes issue ASTERISK-18969)
Reported by: rmudgett
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1656/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350364 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The time passed by the LOCK function to an internal function was relative
time when the function expected absolute time.
* Don't use C++ keywords in get_lock().
(closes issue ASTERISK-16868)
Reported by: Andrey Solovyev
Patches:
20101102__issue18207.diff.txt (license #5003) patch uploaded by Andrey Solovyev (modified)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The iax2_process_thread() can exit without anyone waiting to join the
thread. If noone is waiting to join the thread then a large memory leak
occurs.
* Made iax2_process_thread() deatach itself if nobody is waiting to join
the thread.
(closes issue ASTERISK-17339)
Reported by: Tzafrir Cohen
Patches:
asterisk-1.8.4.4-chan_iax2-detach-thread-on-non-stop-exit.patch (license #5617) patch uploaded by Alex Villacis Lasso (modified)
(closes issue ASTERISK-17825)
Reported by: wangjin
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350220 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The Asterisk -x command line parameter is documented inconsistently.
* Made the -x documentation and behavior consistent.
* Since this is also a new year, updated the copyright notices while here.
(closes issue ASTERISK-19094)
Reported by: Eugene
Patches:
issueA19094_correct_asterisk_option_x.patch (license #5674) patch uploaded by Walter Doekes (modified)
Tested by: Eugene
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350075 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350023 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
(closes issue ASTERISK-16538)
Review: https://reviewboard.asterisk.org/r/1649/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@349968 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
(closes issue ASTERISK-18963)
Reported by: socketpair
Patches:
fix.patch (license #6337) patch uploaded by socketpair
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@349819 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@349731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@349728 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@349672 65c4cc65-6c06-0410-ace0-fbb531ad65f3
