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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@210302 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r210237 | kpfleming | 2009-08-04 09:51:39 -0500 (Tue, 04 Aug 2009) | 10 lines
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.
(closes issue #15606)
Reported by: mvanbaak
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@210238 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@210190 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@210154 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209839 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r209759 | kpfleming | 2009-07-31 19:52:00 -0500 (Fri, 31 Jul 2009) | 7 lines
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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209760 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
ABE-1822
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209673 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209279 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209256 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r209131 | mmichelson | 2009-07-27 12:44:06 -0500 (Mon, 27 Jul 2009) | 18 lines
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.
(closes issue #15182)
Reported by: CGMChris
Patches:
15182.patch uploaded by mmichelson (license 60)
Tested by: CGMChris
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209132 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209056 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r208990 | mvanbaak | 2009-07-27 11:56:13 +0200 (Mon, 27 Jul 2009) | 5 lines
backport rev 205532 from trunk:
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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208991 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Requiring 'module reload' to reload everything, including
core etc makes russell very unhappy.
The default configuration already loads the 'friendly' aliases template.
Added 'reload=module reload' to that template.
Also removed the comment in main/cli.c that reload should come back.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208813 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r208746 | jpeeler | 2009-07-25 01:19:50 -0500 (Sat, 25 Jul 2009) | 7 lines
Fix compiling under dev-mode with gcc 4.4.0.
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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208749 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208706 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r208622 | mmichelson | 2009-07-24 14:24:28 -0500 (Fri, 24 Jul 2009) | 16 lines
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.
(closes issue #15559)
Reported by: amorsen
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208630 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r208592 | russell | 2009-07-24 13:38:24 -0500 (Fri, 24 Jul 2009) | 7 lines
Do not log an ERROR if autoservice_stop() returns -1.
This does not indicate an error. A return of -1 just means that the channel
has been hung up.
(reported in #asterisk-dev)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208593 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r208587 | mmichelson | 2009-07-24 13:26:50 -0500 (Fri, 24 Jul 2009) | 10 lines
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.
(closes issue #14575)
Reported by: chris-mac
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208588 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208548 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The function to check wether we need to install packages was using
dpkg-query which was gives wrong output on Debian 5
Also, the apt-get has been replaced with aptitude because aptitude
is now the preferred way to handle packages on Debian
(closes issue #15570)
Reported by: mvanbaak
Patches:
2009072400_installprereq-aptitude.diff uploaded by mvanbaak (license 7)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208542 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
Review: https://reviewboard.asterisk.org/r/310/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208464 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Jeff Peeler
Kevin P. Fleming
Richard Mudgett
Tilghman Lesher
Mark Michelson
Matthew Nicholson
Bradley Latus
Russell Bryant
Sean Bright
David Brooks
Michiel van Baak