The iax frame's retrans sched id was written over right
before iax2_frame_free was called. In iax2_frame_free that
retrans id is used to delete the sched item. By writing over
the retrans field before the sched item could be deleted, it was
possible for a retransmit to occur on a freed frame.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@219519 65c4cc65-6c06-0410-ace0-fbb531ad65f3
INVITE requests must always contain a new unique branch id. When
a new branch id is created for an INVITE, the dialog's invite_branch
variable must be updated so CANCEL requests use the correct branch id.
(closes issue #15262)
Reported by: maniax
Patches:
asterisk-1.6.1.0-sip-branch.patch uploaded by tweety (license 608)
invite_new_branch_trunk.diff uploaded by dvossel (license 671)
Tested by: maniax, dvossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@219450 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch cleans up the locking logic in chan_sip.c's
handle_invite_replaces() function as well as making use
of ast_do_masquerade() rather than forcing the masquerade
on an ast_read(). The code had several redundant unlocks
that would result in 'freed more times than we've locked!'
errors. I cleaned these up as well as moving all the unlock
logic to the end of the function. This patch should also
resolve the issue people were having with the replacecall
channel never being unlocked with one legged calls.
(closes issue #15151)
Reported by: irroot
Patches:
invite_w_replaces_1.4.diff uploaded by dvossel (license 671)
Tested by: irroot, dvossel
Review: https://reviewboard.asterisk.org/r/371/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@219303 65c4cc65-6c06-0410-ace0-fbb531ad65f3
After talking to rmudgett about some of his recent iflist locking changes, it
was determined that the only place that would destroy a channel without being
explicitly to do so was in handle_init_event. The loop to walk the interface
list has been modified to wait to destroy the channel until the dahdi_pvt of
the channel to be destroyed is no longer needed.
(closes issue #15378)
Reported by: samy
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@218401 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The IAX2 Call Token security patch inadvertently broke the use of
encryption due to the reorganization of code in the socket_process()
function. When encryption is used, an incoming full frame must first
be decrypted before the information elements can be parsed. The
security release mistakenly moved IE parsing before decryption in
order to process the new Call Token IE. To resolve this, decryption
of full frames is once again done before looking into the frame. This
involves searching for an existing callno, checking the pvt to see if
encryption is turned on, and decrypting the packet before the internal
fields of the full frame are accessed.
associated with AST-2009-006
(closes issue #15834)
Reported by: karesmakro
Patches:
iax2_encryption_fix_1.4.diff uploaded by dvossel (license 671)
Tested by: dvossel, karesmakro
Review: https://reviewboard.asterisk.org/r/355/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@217806 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Remove code that causes loops in registrations.
We have agreed that the patch that this code was part of was bad. I am ripping out the code that causes
the issue. putnopvut needs to check the rest of the patch, if it needs to be changed as well.
This solves the issue reported in #15540, but needs more work before we close it (as described above).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@217668 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI
links *before* any call progress. The SIP channel receives these frames and by default
signals 183 Session progress and starts sending media. This will cause phones to
play silence and ignore the later 180 ringing message. A bad user experience.
The fix is twofold:
- We discovered that asterisk apps that support early media ("noanswer") did not send
any PROGRESS frame to indicate early media. Fixed.
- We introduce a setting in chan_sip so that users can disable any relay of media frames
before the outbound channel actually indicates any sort of call progress.
In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions
of Asterisk, this will be enabled. We don't assume that it will change your Asterisk
phone experience - only for the better.
We encourage third-party application developers to make sure that if they have applications
that wants to send early media, add a PROGRESS control frame transmission to make sure that
all channel drivers actually will start sending early media. This has not been the default
in Asterisk previous to this patch, so if you got inspiration from our code, you need to
update accordingly. Sorry for the trouble and thanks for your support.
This code has been running for a few months in a large scale installation (over 250
servers with PRI and/or BRI links to old PBX systems).
That's no proof that this is an excellent patch, but, well, it's tested :-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@216430 65c4cc65-6c06-0410-ace0-fbb531ad65f3
From section 13.3.1.1 of RFC 3261:
If the UAS desires an extended period of time to answer the INVITE,
it will need to ask for an "extension" in order to prevent proxies
from canceling the transaction. A proxy has the option of canceling
a transaction when there is a gap of 3 minutes between responses in a
transaction. To prevent cancellation, the UAS MUST send a non-100
provisional response at every minute, to handle the possibility of
lost provisional responses.
(closes issue #11157)
Reported by: rjain
Tested by: twilson
Review: https://reviewboard.asterisk.org/r/315/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@215682 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If more ports were specified than configured in misdn.conf a reload would crash
asterisk. The problem was the unconfigured port was using data from the
previously configured port. When the data for an unconfigured port was freed a
crash would result from the double free.
(closes issue #12113)
Reported by: agupta
Patches:
bug12113.patch uploaded by jpeeler (license 325)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@212498 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There already was code present to be sure that a CANCEL will contain the same branch-id
as the INVITE it is cancelling. However, for INVITES which are challenged downstream,
this mechanism did not work properly. Now this is taken care of.
This is a backport of a fix already present in all 1.6.X branches and in trunk. It also
fixes ABE-1907.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@211807 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Issue 15655: For the case where dialing is complete for an incoming
call, dahdi_new() was asked to start the PBX and then the code set more
channel variables. If the dialplan hungup before these channel variables
got set, asterisk would likely crash.
* Fixed potential for overlap incoming call to erroneously set channel
variables as global dialplan variables if the ast_channel structure failed
to get allocated.
* Added missing set of CALLINGSUBADDR in the dialing is complete case.
(closes issue #15655)
Reported by: alecdavis
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@210575 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Prior to this patch, a wildcard extension in the dialplan (for example, _*.) would take
precedence over picking up a call in the channel's pickup group. This patch simply moves
the block of code handling pickup group matching to above the extension matching code.
(closes issue #14735)
Reported by: stevedavies
Review: https://reviewboard.asterisk.org/r/319/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@210067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The latest GCC (what will become 4.5.x) has a few new warnings, that in these
cases found some either downright buggy code, or at least seriously poorly
designed code that could be improved.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@209759 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Mostly trivial changes, but I did not know of any other way to fix the
"dereferencing type-punned pointer will break strict-aliasing rules" error
without creating a tmp variable in chan_skinny.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@208746 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For cases where Asterisk sends an INVITE and receives a non 2XX final
response, Asterisk would follow the INVITE transaction by immediately
sending a BYE, which was unnecessary.
(closes issue #14575)
Reported by: chris-mac
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@208587 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This generalizes the fix for issue 13849. The initial fix corrected the
problem that Asterisk would reply with a 491 if a reinvite were received
from an endpoint and we had not yet received an ACK from that endpoint
for the initial INVITE it had sent us. This expansion also allows Asterisk
to appropriately handle an INVITE with authorization credentials if Asterisk
had not received an ACK from the previous transaction in which Asterisk had
responded to an unauthorized INVITE with a 407.
(closes issue #14239)
Reported by: klaus3000
Patches:
14239.patch uploaded by mmichelson (license 60)
Tested by: klaus3000
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@208386 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There was already code for other signaling types in dahdi_handle_event to
handle dialing if a dial operation dial string was present. Simply add
SIG_EMWINK to the list.
(closes issue #14434)
Reported by: araasch
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@207827 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit changes the build system so that user-provided flags (in ASTCFLAGS
and ASTLDFLAGS) are supplied to the compiler/linker *after* all flags provided
by the build system itself, so that the user can effectively override the
build system's flags if desired. In addition, ASTCFLAGS and ASTLDFLAGS can now
be provided *either* in the environment before running 'make', or as variable
assignments on the 'make' command line. As a result, the use of COPTS and LDOPTS
is no longer necessary, so they are no longer documented, but are still supported
so as not to break existing build systems that supply them when building Asterisk.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@207647 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds a new dahdi_wait function to specifically wait for the wink
event. If the wink is not eventually received the channel is hung up.
(closes issue #14434)
Reported by: araasch
Patches:
emwinkmod uploaded by araasch (license 693)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@207573 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Copied from Review board:
In issue 12434, the reporter describes a situation in which audio and video
is offered on the call, but because videosupport is disabled in sip.conf,
Asterisk gives no response at all to the video offer. According to RFC 3264,
all media offers should have a corresponding answer. For offers we do not
intend to actually reply to with meaningful values, we should still reply
with the port for the media stream set to 0.
In this patch, we take note of what types of media have been offered and
save the information on the sip_pvt. The SDP in the response will take into
account whether media was offered. If we are not otherwise going to answer
a media offer, we will insert an appropriate m= line with the port set to 0.
It is important to note that this patch is pretty much a bandage being
applied to a broken bone. The patch *only* helps for situations where video
is offered but videosupport is disabled and when udptl_pt is disabled but
T.38 is offered. Asterisk is not guaranteed to respond to every media offer.
Notable cases are when multiple streams of the same type are offered.
The 2 media stream limit is still present with this patch, too.
In trunk and the 1.6.X branches, things will be a bit different since Asterisk
also supports text in SDPs as well.
(closes issue #12434)
Reported by: mnnojd
Review: https://reviewboard.asterisk.org/r/311
Review: https://reviewboard.asterisk.org/r/313
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@207423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Previously overlap dialing could only be turned on or off for both incoming and
outgoing calls. New parameters incoming, outgoing, and both have been added to
allow further control. There is no change in default behavior with these new
options and allows in band DTMF to be accepted in one direction if required.
(closes issue #14471)
Reported by: eboscani
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@207092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Some ITSP make use of the "Anonymous" display name to detect a
requirement to withhold caller id across the PSTN. This does
not work if the display name is "Unknown".
(closes issue #14465)
Reported by: Nick_Lewis
Patches:
chan_sip.c-callerpres.patch uploaded by Nick (license 657)
chan_sip.c-callerpres_trunk.patch uploaded by dvossel (license 671)
Tested by: Nick_Lewis, dvossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@206938 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/be/branches/C.2-...
..........
Fixed chan_misdn crash because mISDNuser library is not thread safe.
With Asterisk the mISDNuser library is driven by two threads concurrently:
1. channels/misdn/isdn_lib.c::manager_event_handler()
2. channels/misdn/isdn_lib.c::misdn_lib_isdn_event_catcher()
Calls into the library are done concurrently and recursively from
isdn_lib.c.
Both threads can fiddle with the master/child layer3_proc_t lists. One
thread may traverse the list when the other interrupts it and then removes
the list element which the first thread was currently handling. This is
exactly what caused the crash. About 60 calls were needed to a Gigaset
CX475 before it occurred once.
This patch adds locking when calling into the mISDNuser library.
This also fixes some cb_log calls with wrong port parameter.
JIRA ABE-1913
Patches: misdn-locking.patch (Modified with mostly cosmetic changes)
..........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@206706 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* issue #14355 - Crash if attempt to transfer a call to an application.
Masquerade the other pair of the four asterisk channels involved in the
two calls. The held call already must be a bridged call (not an
applicaton) or it would have been rejected.
* issue #14692 - Held calls are not automatically cleared after transfer.
Allow the core to initate disconnect of held calls to the ISDN port. This
also fixes a similar case where the party on hold hangs up before being
transferred or taken off hold.
* JIRA ABE-1903 - Orphaned held calls left in music-on-hold.
Do not simply block passing the hangup event on held calls to asterisk
core.
* Fixed to allow held calls to be transferred to ringing calls.
Previously, held calls could only be transferred to connected calls.
* Eliminated unused call states to simplify hangup code.
* Eliminated most uses of "holded" because it is not a word.
(closes issue #14355)
(closes issue #14692)
Reported by: sodom
Patches:
misdn_xfer_v14_r205839.patch uploaded by rmudgett (license 664)
Tested by: rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@206487 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r206384 | russell | 2009-07-14 09:45:47 -0500 (Tue, 14 Jul 2009) | 6 lines
Ensure apathetic replies are sent out on the proper socket.
chan_iax2 supports multiple address bindings. The send_apathetic_reply()
function did not attempt to send its response on the same socket that the
incoming message came in on.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@206385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
From the review board request:
The fix from review 298 has exposed a new bug in chan_sip.
When we hang up an outgoing call, we first will dump all the outstanding
packets on the sip_pvt using __sip_pretend_ack. Then, if we can, we send
a CANCEL. The problem with this is that since destroyed all the outstanding
packets on the dialog, we cannot match the incoming 487 response to our
INVITE. Because we cannot match the response, we do not send an ACK.
To correct this, instead of using __sip_pretend_ack, I have changed the code
to loop through the list of packets and call __sip_semi_ack on each one
instead. This causes us to stop retransmitting the requests, but we still have
them around in case we get responses for them.
When discussing this earlier today with Josh Colp, we both agreed that in the
majority of cases, this would be enough of a fix. However, we also agreed that
we should have a safety net in place in case we never receive a response to
our initial INVITE. To handle this, I have modified __sip_autodestruct to
behave similar to the way it does in Asterisk 1.4. If there are outstanding
packets on the sip_pvt, the needdestroy flag is not set, and the last request
on the dialog was either a CANCEL or BYE, then we set the needdestroy flag and
reschedule destruction for 10 seconds in the future. If, though, the
needdestroy flag is set, then we use __sip_pretend_ack to kill the remaining
outstanding packets so that the monitor thread can destroy the sip_pvt.
I ran two separate tests. First, I placed a call from my Aastra phone to my
Polycom phone. I hung up the Aastra before the Polycom answered. I verified
through sip debug output that Asterisk properly ACKed the 487 received from the
Polycom.
For my second test, I set up a SIPp UAS scenario so that it would send a 200 OK
in response to a CANCEL but would not send a 487 for the original INVITE. I
verified that after about 40 seconds, Asterisk properly cleans up the outgoing
sip_pvt for the call.
Review: https://reviewboard.asterisk.org/r/308
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@205877 65c4cc65-6c06-0410-ace0-fbb531ad65f3
