If a BLF subscription exists for long enough, using %d may print negative version numbers.
Unlikely, as 2^32 at 1 update per second is ~137 years, or half that before the versions number started going negative.
Tested with Asterisk 1.8.8.2 with Grandstream phones.
alecdavis (license 585)
Tested by: alecdavis
Review: https://reviewboard.asterisk.org/r/1694/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352862 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For whatever reason, we don't have a single function for copying data like this
from SIP peers to the SIP pvt. This patch adds the copying of amaflags to the
sip_pvt, but it would probably be worth discussing this function along with
the others that essentially just copy some amount of data from a peer to a
private.
(Closes issue ASTERISK-19029)
Reported by: Matt Lehner
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352755 65c4cc65-6c06-0410-ace0-fbb531ad65f3
main/test.c includes "asterisk/version.h", when it should include
"asterisk/ast_version.h" instead (and it should use the ast_get_version()
and ast_get_version_num() functions). This commit modifies it to extract
the Asterisk version information using the proper APIs, and as a result means
that main/test.c no longer needs to be rebuilt when a Subversion checkout
is updated or modified.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352612 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There was faulty information in the sample config describing user as a synonym for friend
so it has been changed to better elaborate on the differences between the three entity
types.
(closes issue ASTERISK-15537)
Reported by: yarique
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352511 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Now that we have the right license for the Russian 1.4.22 sounds as well as the sounds
for the Australian English 1.4.22 sounds, we can finally set the sounds to use 1.4.22!
(closes issue ASTERISK-18978)
Reported by: Cameron Twomey
Patches:
confbridge.tar.001 uploaded by Cameron Twomey
confbridge.tar.002 uploaded by Cameron Twomey
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352367 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This prevents the 'h' extension from being run on the transferee
channel when it is transferred via a native transfer mechanism such
as SIP REFER.
(closes ASTERISK-19173)
Reported by: Ross Beer
Tested by: Kristjan Vrban
Patches:
ASTERISK-19173 by Mark Michelson (license 5049)
Review: https://reviewboard.asterisk.org/r/1685
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352199 65c4cc65-6c06-0410-ace0-fbb531ad65f3
While the FAXOPT function could be used to set the modem capabilities, the
input to that function was not being applied correctly to the spandsp layer.
This patch applies the current model capabilities before starting the spandsp
layer.
(closes issue: ASTERISK-16409)
Reported by: Kristijan Vrban
Tested by: Matt Jordan, Matthew Nicholson
Patches:
spandsp-modems-1.8.diff uploaded by mnicholson (license 5081)
spandsp-modems-10.diff uploaded by mnicholson (license 5081)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352144 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The invalid value used when notifycid was enabled was benign. As far as
the code was concerned -1 and 1 are equivalent.
(closes issue ASTERISK-19232)
Reported by: Eike Kuiper
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352090 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Note: Noone calls ast_app_dtget() with the timeout parameter of zero so
the bad code normally will never get executed.
* Fix unnecessary floating point division in func_timeout.c
timeout_write() when all other values are integers.
(closes issue ASTERISK-16817)
Reported by: Dmitry Andrianov
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352029 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
