The configuration option for allowing hosts to make non-token-based calls
is 'calltokenoptional', not 'calltokenignore'. (reported on asterisk-users)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@230246 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Crash related to chan_misdn connection. Patch submitted by gknispel_proformatique, tested
by francesco_r. "I have many crash since i have upgraded to Asterisk 1.4.27-rc2. Attached
a full bt." This patch zeros out an ast_frame.
(closes issue #16041)
Reported by: francesco_r
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@228078 65c4cc65-6c06-0410-ace0-fbb531ad65f3
With the new code, media level proprieties should no longer be confused with session level proprieties. This change also reorganizes some of the SDP parsing code which should make it easier to manage in the future.
(closes issue #14994)
Reported by: frawd
Tested by: frawd, mnicholson, file
Review: https://reviewboard.asterisk.org/r/385/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@227758 65c4cc65-6c06-0410-ace0-fbb531ad65f3
SIP channel names were supposed to be unique by way of a name suffix derived from the
pointer to the channel's private data. Uniqueness was preserved on 32-bit systems, but
not on 64-bit systems. This patch, as suggested by kpfleming, replaces this suffix with
a simple incremented unsigned int.
(closes issue #15152)
Reported by: palbrecht
Review: https://reviewboard.asterisk.org/r/420/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@226972 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It is possible for the PBX thread to queue up signaling frames before
a destination call number is received. This can result in signaling
frames being sent out with no destination call number. Since recent
versions of Asterisk require accurate destination callnumbers for all
Full Frames, this can cause a VNAK loop to occur. To resolve this
no signaling frames are sent until a destination callnumber is received,
and destination call numbers are now only required for iax_pvt matching
when the frame is an ACK.
Review: https://reviewboard.asterisk.org/r/413/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@225243 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The shrinking of caller id removes '(', ' ', ')', non-trailing '.',
and '-' from the string. This means values such as 555.5555 and
test-test result in 555555 and testtest. There are instances,
such as Skype integration, where a specific value is passed via
caller id that must be preserved unmodified. This patch makes
the shrinking of caller id optional in chan_sip and chan_iax in
order to support such cases. By default this option is on to
preserve previous expected behavior.
(closes issue #15940)
Reported by: dimas
Patches:
v2-15940.patch uploaded by dimas (license 88)
15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671)
Tested by: dvossel
Review: https://reviewboard.asterisk.org/r/408/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@225032 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The problem here is that chan_dahdi is designed in such a way to set
certain values in the dahdi_pvt only once. One of those such values
is the configured caller id data in chan_dahdi.conf. For PRI, the
configured caller id data could be overwritten during a call. Instead
of saving the data and restoring, it was decided that for all non-analog
channels it was simply best to not set the configured caller id in the
first place and also clear it at the end of the call.
(closes issue #15883)
Reported by: jsmith
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@224330 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When the Busy() or Congestion() application is used towards ISDN (an ISDN
progress is sent), the responding ISDN Disconnect or Release may contain
the ISDN cause user busy or one of the congestion causes. In chan_dahdi.c
these causes will only set the needbusy or needcongestion flags and not
activate the softhangup procedure. Unfortunately only the latter can
interrupt the endless wait loop of Busy()/Congestion().
Result: PRI channels staying in state busy for the rest of asterisk life
or until the other end times out and forces the call to clear.
(in issue 0014292)
Reported by: tomaso
Patches:
disc_rel_userbusy.patch uploaded by tomaso (license 564)
(This patch is unrelated to the issue.)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@224260 65c4cc65-6c06-0410-ace0-fbb531ad65f3
chan_sip has some code to automatically switch from T.38 mode to voice mode when
a voice frame is written to the channel while it is in T.38 mode; this was
intended to handle the situation when a FAX transmission has ended and the channel
is not yet hung up, but is causing problems at the beginning of FAX sessions as
well when there are still voice frames 'in flight' at the time the T.38 negotiation
completes. This patch removes the automatic switchover.
(issue #16025)
Reported by: jamicque
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@223692 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Memory leak when the same config option is set more than once in an
misdn.conf section. Why must this be considered? Templates! Defining a
template with default port options and later adding to or overriding some
of them.
Patches:
memleak-misdn.patch
JIRA ABE-1998
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@222797 65c4cc65-6c06-0410-ace0-fbb531ad65f3
misdn.conf: astdtmf must be set to "yes". With "no", buffer loss does not
occur.
The translated frame "f2" when passing through ast_dsp_process() is not
freed whenever it is not used further in process_ast_dsp(). Then in the
end it is never ever freed.
Patches:
translate.patch
JIRA ABE-1993
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@222691 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The variable index used in this scenario for accessing the dahdi_pvts was
wrong and was most likely copied from the several other places it is used
correctly.
(closes issue #15998)
Reported by: tsearle
Patches:
dahdi_reset_crash.patch uploaded by tsearle (license 373)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@222393 65c4cc65-6c06-0410-ace0-fbb531ad65f3
See Mantis issue for details of what prompted this change.
Additional notes:
This patch changes the ao2_iterator API in two ways: F_AO2I_DONTLOCK
has become an enum instead of a macro, with a name that fits our
naming policy; also, it is now necessary to call
ao2_iterator_destroy() on any iterator that has been
created. Currently this only releases the reference to the container
being iterated, but in the future this could also release other
resources used by the iterator, if the iterator implementation changes
to use additional resources.
(closes issue #15987)
Reported by: kpfleming
Review: https://reviewboard.asterisk.org/r/383/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@222152 65c4cc65-6c06-0410-ace0-fbb531ad65f3
I have not been able to reproduce the problem of losing channels.
However, I have seen in the code a reentrancy problem that might give
these symptoms.
The reentrancy patch does several things:
1) Guards B channel and B channel structure allocation.
2) Makes the B channel structure find routines more precise in locating records.
3) Never leave a B channel allocated if we received cause 44.
The last item may cause temporary outgoing call problems, but they should
clear when the line becomes idle.
(closes issue #15490)
Reported by: slutec18
Patches:
issue15490_channel_alloc_reentrancy.patch uploaded by rmudgett (license 664)
Tested by: rmudgett, slutec18
(closes issue #15458)
Reported by: FabienToune
Patches:
issue15458_channel_alloc_reentrancy.patch uploaded by rmudgett (license 664)
Tested by: FabienToune, rmudgett, slutec18
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@221769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds a new field "portinuri" to the sip dialog struct and the sip peer struct. That field is used during RURI generation to determine if the port should be included in the RURI. It is also used in some places to determine if an SRV lookup should occur.
(closes issue #14418)
Reported by: klaus3000
Tested by: klaus3000, mnicholson
Review: https://reviewboard.asterisk.org/r/369/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@221360 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Be default, change SSRC when doing an audio stream changes Asterisk doesn't
honor marker bit when reinvited to already-bridged RTP streams,resulting in
far-end stack discarding packets with "old" timestamps that areactually part of
a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is a
reinvite, unless the 'constantssrc' is set to true in sip.conf.
The original issue reported to Digium support detailed the following situation:
ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in
fromITSP, Asterisk dials the app server which sends a re-invite back
toAsterisk--not to negotiate to send media directly to the ITSP, but to
indicatethat it's changing the stream it's sending to Asterisk. The app
servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker
bit on the new stream. Asterisk passes through the teimstamp of the new stream,
butdoes not reset the SSRC, sequence numbers, or set the marker bit.
When the timestamp on the new stream is older than the timestamp on the
originalstream, the ITSP (which doesn't know there has been any change) discards
the newframes because it thinks they are too old. This patch addresses this by
changing the SSRC on a stream update unless constantssrc=true is set in
sip.conf.
Review: https://reviewboard.asterisk.org/r/374/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@221086 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This fixes a 100% CPU problem in the SIP driver, found by profiling
the driver while the problem was occurring.
(closes issue #14309)
Reported by: pkempgen
Patches:
20090924__issue14309.diff.txt uploaded by tilghman (license 14)
Tested by: pkempgen, vrban
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@220873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
If an endpoint sends two registration requests in a very short
period of time with the same nonce, both receive 401 responses
from Asterisk, each with a different nonce (the second 401
containing the current nonce and the first one being stale).
If the endpoint responds to the first 401, it does not match
the current nonce so Asterisk sends a third 401 with a newly
generated nonce (which updates the current nonce)... Now if
the endpoint responds to the second 401, it does not match the
current nonce either and Asterisk sends a fourth 401 with a
newly generated nonce... This loop goes on and on.
There appears to be a simple fix for this. If the nonce from
the request does not match our nonce, but is a good response
to a previous nonce, instead of sending a 401 with a newly
generated nonce, use the current one instead. This breaks
the loop as the nonce is not updated until a response is
received. Additional logic has been added to make sure no
nonce can be responded to twice though.
(closes issue #15102)
Reported by: Jamuel
Patches:
patch-bug_0015102 uploaded by Jamuel (license 809)
nonce_sip.diff uploaded by dvossel (license 671)
Tested by: Jamuel
Review: https://reviewboard.asterisk.org/r/289/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@205804 65c4cc65-6c06-0410-ace0-fbb531ad65f3
With this change, we make note of Record-Route headers present in any SUBSCRIBE
request that we receive so that our outbound NOTIFY requests will have the proper
Route headers in them.
(closes issue #14725)
Reported by: ibc
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@205775 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Add missing clearing of the dialing flag when the ISDN call is CONNECTED.
(i.e. When libpri generates the event PRI_EVENT_ANSWER.)
(closes issue #15420)
Reported by: scottbmilne
Patches:
bug15420-1.4.25.1-diff2.txt uploaded by alecdavis (license 585)
Tested by: scottbmilne, alecdavis
(closes issue #15416)
Reported by: avinoash
(closes issue #15389)
Reported by: alecdavis
This patch should also fix the following issue:
(issue #15205)
Reported by: vinsik
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@205728 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If an incoming mISDN call is answered with the Answer application and a
subsequent Dial gets a busy endpoint then it is valid for that already
connected channel to get the busy indication. Asterisk will play the busy
tones until the dialplan plays something else or hangs up the call.
(closes issue #11974)
Reported by: fvdb
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@204834 65c4cc65-6c06-0410-ace0-fbb531ad65f3
a DNS lookup fails on a host or outboundproxy.
(closes issue #13432)
Reported by: p_lindheimer
Patches:
outboundproxy.patch uploaded by p (license 558)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@204300 65c4cc65-6c06-0410-ace0-fbb531ad65f3
chan_sip has had a problem for quite a long time that would manifest when
Asterisk would send multiple SIP responses on the same dialog before receiving
a response. The problem occurred because chan_sip only kept track of the highest
outgoing sequence number used on the dialog. If Asterisk sent two requests out,
and a response arrived for the first request sent, then Asterisk would ignore
the response. The result was that Asterisk would continue retransmitting the
requests and ignoring the responses until the maximum number of retransmissions
had been reached.
The fix here is to rearrange the code a bit so that instead of simply comparing
the sequence number of the response to our latest outgoing sequence number, we
walk our list of outstanding packets and determine if there is a match. If there is,
we continue. If not, then we ignore the response.
In doing this, I found a few completely useless variables that I have now removed.
(closes issue #11231)
Reported by: flefoll
Review: https://reviewboard.asterisk.org/r/298
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@204243 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Using the SPINLOCK schedule delete macro can result in the iax_pvt lock
being given up. This makes it possible for the iax_pvt to dissappear
when we thought we held the mutex the entire time. To resolve this, the
iax_pvt's ref count is incremented.
(closes issue #15377)
Reported by: aragon
Patches:
iax_spin_issue_1.4.diff uploaded by dvossel (license 671)
Tested by: aragon, dvossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@204067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Before this patch, Asterisk unconditionally picked B channels exclusively
on the CPE side and normally allowed alternative B channels on the network
side. Now Asterisk does the opposite.
Reasons for the CPE side to normally not pick B channels exclusively:
* For CPE point-to-multipoint mode (i.e. phone side), the CPE side does
not have enough information to exclusively pick B channels. (There may be
other devices on the line.)
* Q.931 gives preference to the network side picking B channels.
* Some telcos require the CPE side to not pick B channels exclusively.
(closes issue #14383)
Reported by: mbrancaleoni
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@203908 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change resolves a crash observed locally during some T.38 testing.
A call was set up using a call file, and when the T.38 reinvite came in,
the channel state was still AST_STATE_DOWN. The reason is explained by
a comment in the code that previously lived in the handling of
AST_STATE_RINGING. This change modifies the logic to handle the same
race condition for any channel state that is not UP.
(closes ABE-1895)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@203115 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Valid format is: pritimer=timer_name,timer_value
* Fixed segfault if the ',' is missing.
* Completely check the range returned by pri_timer2idx() to prevent
possible access outside array bounds.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@203036 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If a CANCEL were received by Asterisk, we would send a 487 in response
to the original INVITE and a 200 OK for the CANCEL. If there were a network
hiccup which caused the 200 OK and the 487 to be lost, then the UA communicating
with Asterisk may try to retransmit its CANCEL. Asterisk's response to this used
to be to try sending another 487 to the canceled INVITE and another 200 OK to the
CANCEL.
The problem here is that the originally-sent 487 was sent "reliably" meaning that
it will be retransmitted until it is received properly. So when we receive the second
CANCEL it is likely that the first batch of 487s we sent is still going strong and
reaches the UA. The result was that the second set of 487s would be retransmitted
constantly until the maximum number of retries had been reached.
The fix for this is that if we receive a second CANCEL for an INVITE, then we cancel
the retransmission of the first set of 487s and start a second set. This causes the
dialog to be terminated reasonably.
(closes issue #14584)
Reported by: klaus3000
Patches:
14584_v2.patch uploaded by mmichelson (license 60)
Tested by: klaus3000
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@202341 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There was a while loop in get_ip_and_port_from_sdp which was controlled
by a call to get_sdp_iterate. The loop would exit either if what we were
searching for was found or if the return was NULL. The problem is that
get_sdp_iterate never returns NULL. This means that if what we were searching
for was not present, the loop would run infinitely. This modification of the
loop fixes the problem.
(closes issue #15213)
Reported by: schmidts
(closes issue #15349)
Reported by: samy
(closes issue #14464)
Reported by: pj
(closes issue #15345)
Reported by: aragon
Patches:
sip_inf_loop.patch uploaded by mmichelson (license 60)
Tested by: aragon
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@202336 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Zombie channels could be passed, and chan_sip.c wasn't checking for it.
Could crash Asterisk. Now checking for NULL pointer.
(closes issue #15330)
Reported by: okrief
Tested by: dbrooks
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@201380 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When we receive a SIP reinvite, it is possible that we may not be able to process the
reinvite immediately since we have also sent a reinvite out ourselves. The problem is
that whoever sent us the reinvite may have also sent a reinvite out to another party,
and that reinvite may have succeeded.
As a result, even though we are not going to accept the reinvite we just received, it
is important for us to not have problems if we suddenly start receiving RTP from a new
source. The fix for this is to grab the media source information from the SDP of the
reinvite that we receive. This information is passed to the RTP layer so that it will
know about the alternate source for media.
Review: https://reviewboard.asterisk.org/r/252
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@197588 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If we already have an address for a peer, and we are reloading the sip
configuration, try to use that address to contact the peer, instead of
getting it from the Contact.
(closes issue #15194)
Reported by: ibc
Patches:
sip.patch uploaded by eliel (license 64)
Tested by: manwe
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@197562 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The presence of rport is now stored as a separate flag. Once the dialog is setup and authenticated
(or it passes through unauthenticated) the proper nat flag is set.
(closes issue #13823)
Reported by: dimas
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@197466 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There are rare cases in which a frame's delivery timestamp is slightly less than the iax2_pvt's offset. This causes the pvt's timestamp to be a small negative number, but since the timestamp value is unsigned it looks like a huge positive number. This patch checks for this negative case and sets the ms to zero. A similar check is already done right below this one in the 'else' statement.
(closes issue #15032)
Reported by: guillecabeza
Patches:
chan_iax2.c.patch_timestamp uploaded by guillecabeza (license 380)
Tested by: guillecabeza
(closes issue #14216)
Reported by: Andrey Sofronov
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@195991 65c4cc65-6c06-0410-ace0-fbb531ad65f3
IAX was not sending REGREJ to terminate invalid registrations. Instead it sent another REGAUTH if the authentication challenge failed. This caused a loop of REGREQ and REGAUTH frames.
(Related to Security fix AST-2009-001)
(closes issue #14867)
Reported by: aragon
Tested by: dvossel
(closes issue #14717)
Reported by: mobeck
Patches:
regauth_loop_update_patch.diff uploaded by dvossel (license 671)
Tested by: dvossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@194873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There is a bug tracker issue where people are reporting "Ghost" channels in their 'iax2 show channels' output. The confusion is caused by channels being listed as "(NONE)" with format "unknown". These are not channels of coarse. They are usually just pending registration or poke requests, but it is confusing output. To help make sense of this I have added two columns to 'iax2 show channels'. One shows the first message which started the transaction, and the second shows the last message sent by either side of the call. This helps diagnose why the entry exists and why it may not go away.
(closes issue #14207)
Reported by: clive18
Review: https://reviewboard.asterisk.org/r/246/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@194557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The loop detection/spiral detection code in chan_sip used the owner
channel's state as a criterion for determining if the incoming INVITE
is a looped request. The problem with this is that the INVITE-handling
code happens in a different thread than the thread that marks the owner
channel as being up. As a result, if a reinvite were to come in very quickly,
say from another Asterisk on the same LAN, it was possible for the reinvite
to arrive before the owner channel had been set to the up state.
This patch corrects the problem by using the invitestate of the sip_pvt
instead, since that can be guaranteed to be set correctly by the time
the reinvite arrives. Since there is a switch statement further in the
INVITE-handling code, the AST_STATE_RINGING state also checks the invitestate
of the sip_pvt in case we should actually be treating the channel as if it were
up already.
(closes issue #12215)
Reported by: jpyle
Patches:
12215_confirmed.patch uploaded by mmichelson (license 60)
Tested by: lmadsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@194484 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The problem was that the hangup code was setting the invitestate too early. The result of
this was that we would always send a CANCEL request, even if it was not an appropriate
time to do so (e.g. we have not yet received a provisional response for our INVITE).
Note that this same fix had been applied to trunk and the 1.6.X branches starting with
revision 155467. This is why you will see this revision being blocked from those places.
AST-216
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@193880 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In the state MISDN_CALLING (i.e. SETUP was sent but no answer has arrived yet),
it is not allowed to clear the call with RELEASE_COMPLETE. It must be
cleared with DISCONNECT. A RELEASE_COMPLETE is only allowed as an answer
to a SETUP. (See Q.931 ch. 5.3.2, 5.3.2.a, 5.3.2.b)
Patches:
chan-misdn-ccstate7.patch uploaded by customer.
JIRA ABE-1862
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@193613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Added the dialed extension and context to the chan_misdn messages warning
that the dialed number cannot be matched in the dialplan.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@193050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If the fullcontact field appears in both the sippeers and the
sipregs table, then during reconstruction of the field, it will
otherwise be doubled.
(closes issue #14754)
Reported by: Alexei Gradinari
Patches:
20090506__bug14754.diff.txt uploaded by tilghman (license 14)
Tested by: lmadsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@192932 65c4cc65-6c06-0410-ace0-fbb531ad65f3
mohinterpret and mohsuggest global variables were not copied over during build_users and build_peers.
(closes issue #14728)
Reported by: dimas
Patches:
v1-14728.patch uploaded by dimas (license 88)
Tested by: dimas, dvossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@192213 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If both sides of a Local channel were hung up at around the same time it was
possible for one thread to destroy the local private structure and have the other thread
immediately try to remove the already freed structure from the local channel list.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@190286 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Previously, packetization settings were ignored and now they are not. A new
config option 'autoframing' has been added to mirror the way chan_sip handles
it. Turning on the autoframing option (available both as a global option or per
peer) overrides the local settings with the remote packetization settings.
Testing was performed with varying packetization levels with the following
codecs: ulaw, alaw, gsm, and g729.
(closes issue #12415)
Reported by: pj
Patches:
2009012200_h323packetization.diff.txt uploaded by mvanbaak (license 7),
modified by me
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@189991 65c4cc65-6c06-0410-ace0-fbb531ad65f3
An agent logs in by calling an extension that calls the AgentLogin app. In agents.conf ackcall=always is set, so when they get a call they have the choice to either acknowledge it or ignore it. autologoff=10 is set as well, so if the agent ignores the call over 10sec one may assume that the agent should be logged out (and in this case hungup on as well), but this was not happening.
(closes issue #14091)
Reported by: evandro
Patches:
autologoff.diff uploaded by dvossel (license 671)
Review: http://reviewboard.digium.com/r/225/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@189203 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit fixes the scenario where an incoming call is authenticated
using a peer entry. Previously the channel name was created using either
the username setting from the sip.conf entry or the IP address that the
call came from. Now the channel name will be created using the peer name
itself. This commit will not change the way the channel name is generated
for users or friends.
(closes issue #14256)
Reported by: Nick_Lewis
Patches:
chan_sip.c-chname.patch uploaded by Nick (license 657)
Tested by: Nick_Lewis, file
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@188946 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When the caller ID is restricted, the expected behavior is for the caller id to be blank. In chan_dahdi, the national prefix is placed onto the callers number even if its restricted (empty) causing the caller id to be the national prefix rather than blank.
(closes issue #13207)
Reported by: shawkris
Patches:
national_prefix.diff uploaded by dvossel (license 671)
Review: http://reviewboard.digium.com/r/220/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@188646 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Kevin P. Fleming
Joshua Colp
David Vossel
Matthew Nicholson
Jason Parker
David Brooks
Richard Mudgett
Olle Johansson
Jeff Peeler
Jean Galarneau
Terry Wilson
Tilghman Lesher
Russell Bryant
Mark Michelson
Michiel van Baak
Sean Bright
Eliel C. Sardanons
Doug Bailey