Because channel information can be accessed outside of the channel thread, we must lock the channel prior to modifying it.
(closes issue #15397)
Reported by: caspy
Patches:
20090714__issue15397.diff.txt uploaded by tilghman (license 14)
Tested by: caspy
........
Richard Mudgett [Thu, 6 Aug 2009 20:15:11 +0000 (20:15 +0000)]
Miscellaneous minor fixes to sig_analog.
* Sanity adjustments to __analog_ss_thread for sig_analog environment.
* Deleted some duplicated code.
* Fixed analog_ss_thread_start passing the wrong pointer.
Joshua Colp [Thu, 6 Aug 2009 17:47:04 +0000 (17:47 +0000)]
Accept additional T.38 reinvites after an initial one has been handled.
Discussion of this subject has yielded that it is not actually acceptable to change
T.38 parameters after the initial reinvite but declining is harsh and can cause the
fax to fail when it may be possible to allow it to continue. This patch changes things
so that additional T.38 reinvites are accepted but parameter changes ignored. This gives
the fax a fighting chance.
This patch makes some small changes to handle watchdog timeouts in a better way,
and also uses a 'cleaner' method of including the spandsp header files.
(closes issue #14769)
Reported by: andrew
Patches:
app_fax-20090406.diff uploaded by andrew (license 240)
v1-14769.patch uploaded by dimas (license 88)
Tested by: freh, deti, caspy, dimas, sgimeno, Dovid
Dialplan starts execution before the channel setup is complete.
* 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.
Update imapstorage.txt documentation.
Updated the imapstorage.txt documentation to reflect that issues with
c-client versions older than 2007 seem to cause crashing issues that
are not seen with more recent versions. Documentation has been updated
to reflect this.
Fixes dialplan wildcard extension taking precedence over call pickup code.
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.
Jeff Peeler [Tue, 4 Aug 2009 15:35:49 +0000 (15:35 +0000)]
Fix broken call pickup
The find_channel_by_group callback was only looking at the channel that was
attempting to make the pickup instead of the other channels in the container.
Eliminate spurious compiler warnings from system headers on *BSD platforms.
Ensure that system headers located in /usr/local/include are actually treated
as system headers by the compiler, and not as local headers which are subject
to warnings from the -Wundef compiler option and others.
Rename 'canreinvite' option to 'directmedia', with backwards compatibility.
It is clear from multiple mailing list, forum, wiki and other sorts of posts
that users don't really understand the effects that the 'canreinvite' config
option actually has, and that in some cases they think that setting it to 'no'
will actually cause various other features (T.38, MOH, etc.) to not work properly,
when in fact this is not the case. This patch changes the proper name of the
option to what it should have been from the beginning ('directmedia'), but
preserves backwards compatibility for existing configurations.
Richard Mudgett [Mon, 3 Aug 2009 18:05:46 +0000 (18:05 +0000)]
Changes from chan_dahdi that did not make it into sig_pri.
* Moved SUPPORT_USERUSER to sig_pri.c
* Fix PRI_DEADLOCK_AVOIDANCE parameter.
* Whitespace changes.
* Added missing unlock in pri_dchannel():PRI_EVENT_RING case.
* Balanced curly braces.
* ast_debug/ast_log changes from chan_dahdi.
* sig_pri_indicate() should default to return -1 if the indication is not
handled.
Russell Bryant [Sat, 1 Aug 2009 11:02:07 +0000 (11:02 +0000)]
Merged revisions 209838 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r209838 | russell | 2009-08-01 05:59:05 -0500 (Sat, 01 Aug 2009) | 13 lines
Modify how Playtones() is used in Milliwatt() to resolve gain issue.
When Milliwatt() was changed internally to use Playtones() so that the proper
tone was used, it introduced a drop in gain in the output signal. So, use
the playtones API directly and specify a volume argument such that the output
matches the gain of the original Milliwatt() code.
(closes issue #15386)
Reported by: rue_mohr
Patches:
issue_15386.rev2.diff uploaded by russell (license 2)
Tested by: rue_mohr
........
Minor changes inspired by testing with latest GCC.
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.
........
Mark Michelson [Fri, 31 Jul 2009 17:55:44 +0000 (17:55 +0000)]
Improve chan_sip's ability to determine what methods should and should not be used in a dialog.
The previous effort here was to store what a peer is capable of receiving by parsing REGISTER
requests from the peer and keeping that information for as long as the registration was active.
The problem with this is that there are a great number of SIP devices which give no indication
of the methods allowed in their REGISTER requests, and it is unreasonable to try to guess what
the device may or may not support. In addition, some SIP devices have been found to claim support
for a specific method, but their handling the method is less than ideal, or they are actually
lying.
With this patch, we now determine what methods a device supports by parsing the Allow header we
receive from them, and we do this with each new dialog. In addition, a configuration option has
been added so that an administrator can essentially blacklist certain methods from being used
with certain peers if the admin knows that support for a specific method is dodgy or nonexistent.
Kevin P. Fleming [Mon, 27 Jul 2009 21:43:36 +0000 (21:43 +0000)]
Cleanup T.38 negotiation changes.
Convert LOG_NOTICE messages about T.38 negotiation in debug level 1 messages,
clean up some looping logic, and correct an improper use of ast_free() for
freeing an ast_frame.
Kevin P. Fleming [Mon, 27 Jul 2009 21:21:43 +0000 (21:21 +0000)]
Make T.38 switchover in ReceiveFAX synchronous.
In receive mode, if the channel that ReceiveFAX is running on supports T.38,
we should *always* attempt to switch T.38, rather than listening for an incoming
CNG tone and only triggering on that. The channel may be using a low-bitrate
codec that distorts the CNG tone, the sending FAX endpoint may not send CNG
at all, or there could be a variety of other reasons that we don't detect it,
but in all those cases if T.38 is available we certainly want to use it.
Allow for UDPTL to use only even-numbered ports if desired.
There are some VoIP providers out there that will not accept SDP
offers with odd numbered UDPTL ports. While it is my personal opinion
that these VoIP providers are misinterpreting RFC 2327, it really is
not a big deal to play along with their silly little games. Of course,
since restricting UDPTL ports to only even numbers reduces the range
of available ports by half, so the option to use only even port numbers
is off by default. A user can enable the behavior by setting
use_even_ports=yes in udptl.conf.
Kevin P. Fleming [Mon, 27 Jul 2009 15:38:59 +0000 (15:38 +0000)]
Restore explicit export of ASTCFLAGS/ASTLDFLAGS and underscore-variants to sub-makes.
During the recent Makefile improvements I made, it seemed the 'make' was
automatically carrying down the ASTCFLAGS/ASTLDFLAGS settings to sub-makes,
so I removed the explict export of them. However, there are some circumstances
where make does this, and some where it does not, so I've brought them back
to ensure they are always exported. I also removed an extraneous double setting
of _ASTLDFLAGS on *BSD platforms.
pthread_self returns a pthread_t which is not an unsigned int on all
pthread implementations. Casting it to an unsigned int fixes compiler warnings.
........
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.
........
Russell Bryant [Fri, 24 Jul 2009 20:54:37 +0000 (20:54 +0000)]
Note that "reload" needs to be added back.
I keep getting annoyed at having to type "module reload" to reload everything,
so I'm adding a note that we need to add "reload" back. "module reload" doesn't
really make sense as the command to reload everything, including the core.
Don't impose an arbitrary limit on member lines in queues.conf
I know what some of you are thinking: "UGH! Mark, why are you using
ast_strdup and ast_free for the string when you can just use ast_strdupa
and let the memory free itself?! Have the bats been chewing on your brain
again?"
Based on past experiences, I don't like using ast_strdupa inside a loop.
It's a good way to potentially exhaust stack space. Also, since this only
happens when reloading queues, I don't think that heap allocations and
frees are going to be a huge problem.
Only send a BYE when hanging up a channel that is up.
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.
Kevin P. Fleming [Fri, 24 Jul 2009 15:02:53 +0000 (15:02 +0000)]
Resolve a T.38 negotiation issue left over from the udptl-updates merge.
The udptl-updates branch that was merged yesterday failed to properly send back
T.38 SDP responses with the correct error correction mode, if the incoming SDP
from the other end caused us to change error correction modes. This patch
corrects that situation.
Kevin P. Fleming [Thu, 23 Jul 2009 21:57:24 +0000 (21:57 +0000)]
Rework of T.38 negotiation and UDPTL API to address interoperability problems
Over the past couple of months, a number of issues with Asterisk
negotiating (and successfully completing) T.38 sessions with various
endpoints have been found. This patch attempts to address many of
them, primarily focused around ensuring that the endpoints'
MaxDatagram size is honored, and in addition by ensuring that T.38
session parameter negotiation is performed correctly according to the
ITU T.38 Recommendation.
The major changes here are:
1) T.38 applications in Asterisk (app_fax) only generate/receive IFP
packets, they do not ever work with UDPTL packets. As a result of
this, they cannot be allowed to generate packets that would overflow
the other endpoints' MaxDatagram size after the UDPTL stack adds any
error correction information. With this patch, the application is told
the maximum *IFP* size it can generate, based on a calculation using
the far end MaxDatagram size and the active error correction mode on
the T.38 session. The same is true for sending *our* MaxDatagram size
to the remote endpoint; it is computed from the value that the
application says it can accept (for a single IFP packet) combined with
the active error correction mode.
2) All treatment of T.38 session parameters as 'capabilities' in
chan_sip has been removed; these parameters are not at all like
audio/video stream capabilities. There are strict rules to follow for
computing an answer to a T.38 offer, and chan_sip now follows those
rules, using the desired parameters from the application (or channel)
that wants to accept the T.38 negotiation.
3) chan_sip now stores and forwards ast_control_t38_parameters
structures for tracking 'our' and 'their' T.38 session parameters;
this greatly simplifies negotiation, especially for pass-through
calls.
4) Since T.38 negotiation without specifying parameters or receiving
the final negotiated parameters is not very worthwhile, the
AST_CONTROL_T38 control frame has been removed. A note has been added
to UPGRADE.txt about this removal, since any out-of-tree applications
that use it will no longer function properly until they are upgraded
to use AST_CONTROL_T38_PARAMETERS.
Fix a problem where a 491 response could be sent out of dialog.
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.
Jeff Peeler [Thu, 23 Jul 2009 15:59:44 +0000 (15:59 +0000)]
Fix sending of interface identifier unconditionally in sig_pri
The wrong logic was being used in chan_dahdi to convert a sig_pri_chan
to the proper libpri channel number. The most significant bit must only
be set only when trunk groups are being used.
Jeff Peeler [Wed, 22 Jul 2009 22:42:33 +0000 (22:42 +0000)]
Reset the fax buffers back to default settings regardless of signaling in use -
Pointed out by Matt F.
Also in the case of not using a signaling module, set the law back to the
default as well.
Jason Parker [Wed, 22 Jul 2009 21:43:57 +0000 (21:43 +0000)]
Restore an int declaration on PPC platforms.
This x is one crafty little bugger...
It was used for 2 different things (one of which was only done on PPC) in 1.4.
One of the uses were removed in trunk, and with it went the declaration.
Clarify documentation on 'realtime update2' to show more than one condition.
(closes issue #15357)
Reported by: snuffy
Patches:
bug_fix_doc_update2.diff uploaded by snuffy (license 35)
(slightly modified by me)
Force an error if a blank is passed to QUOTE (because the documentation states the argument is not optional).
This change makes URIENCODE and QUOTE behave similarly, since the documentation
states that the argument is not optional, for both.
(closes issue #15439)
Reported by: pkempgen
Patches:
20090706__issue15439.diff.txt uploaded by tilghman (license 14)
........
Wait for wink before dialing when using E&M wink signaling
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.
Ensure that user-provided CFLAGS and LDFLAGS are honored.
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.
........
David Vossel [Mon, 20 Jul 2009 20:45:26 +0000 (20:45 +0000)]
reg->username is parsed only once on sip reload
The registration string can contain an expanded user portion of the
form user@domain. This expanded user portion was stored in
reg->username and parsed each time there is a registration refresh.
Now, the domain portion of the user is parsed and stored separately
in the regdomain field.
(closes issue #14331)
Reported by: Nick_Lewis
Patches:
chan_sip.c.domainparse3.patch uploaded by Nick (license 657)
Tested by: Nick_Lewis, dvossel
Answer video SDP offers properly when videosupport is not enabled.
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.
Only do the chan->fdno check in ast_read() in a developer build.
I changed this check to only happen in a dev-mode build. I also added a
comment explaining what is going on. I also made it so that detection of
this situation does not affect ast_read() operation.
Fixed incoming calls being matched to MSNs without type-of-number prefix added.
For an incoming ISDN call the dialed.number is incorrectly matched against
the configured MSNs in misdn.conf. The numbers passed to the dialplan
include the configured prefix for the dialed.number_type, whereas the
check against the configured MSNs (to decide if the call is accepted at
all), is executed without the configured prefix.
e.g., dialed.number = 241168020, TON = national, configured national
prefix is "0". (This is the TON which is used by ISDN providers in the
Netherlands.)
In chan_misdn.c:cb_events() in case EVENT_SETUP the call to
misdn_cfg_is_msn_valid() uses the unnormalized number 241168020, but 57
lines later the call to read_config() adds the prefix, and the
dialed.number is now 0241168020, which is then used in the dialplan.
misdn_cfg_is_msn_valid() must use the normalized number, too.
channels/chan_misdn.c
* Made bearer2str() use allowed_bearers_array[]
* Made use the causes.h defines instead of hardcoded numbers.
* Made use Asterisk presentation indicator values if either of the
mISDN presentation or screen options are negative.
* Updated the misdn_set_opt application option descriptions.
* Renamed the awkward Caller ID presentation misdn_set_opt
application option value not_screened to restricted.
Deprecated the not_screened option value.
channels/misdn/isdn_lib.c
* Made use the causes.h defines instead of hardcoded numbers.
* Fixed some spelling errors and typos.
* Added all defined facility code strings to fac2str().
channels/misdn/isdn_lib.h
* Added doxygen comments to struct misdn_bchannel.
channels/misdn/isdn_lib_intern.h
* Added doxygen comments to struct misdn_stack.
channels/misdn_config.c
configs/misdn.conf.sample
* Updated the mISDN presentation and screen parameter descriptions.
doc/misdn.txt (doc/tex/misdn.tex)
* Updated the misdn_set_opt application option descriptions.
* Fixed some spelling errors and typos.
................
r158010 | rmudgett | 2008-11-19 19:46:09 -0600 (Wed, 19 Nov 2008) | 9 lines
Merged revision 157977 from
https://origsvn.digium.com/svn/asterisk/team/group/issue8824
........
Fixes JIRA ABE-1726
The dial extension could be empty if you are using MISDN_KEYPAD
to control ISDN provider features.
................
projects / thirdparty / asterisk.git / log
