Previously, if an m-line in an SDP offer or answer had a port number of zero,
that line was skipped, and resulted in an 'Unsupported SDP media type...'
warning message. This was misleading, as the media type was not unsupported,
but was ignored because the m-line indicated that the media stream had been
rejected (in an answer) or was not going to be used (in an offer).
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The previous change made the code look for emN and pciN in addition to what
it did originally, which was search for ethN. However, it needed to be looking
for pciN#N, so that's what it does now.
This also moves the memset() to be before every ioctl().
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I also went ahead and took a little time to make sure that the manager value
AMI_SUCCESS was used instead of just return 0 being thrown around everywhere since that's
how we handle this stuff these days.
(closes issue ASTERISK-19249)
Reporter: Jamuel Starkey
Patches:
res_monitor.c-ASTERISK-19249.diff uploaded by Jamuel Starkey (license 5766)
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r352862 | alecdavis | 2012-01-27 13:05:30 +1300 (Fri, 27 Jan 2012) | 12 lines
rfc4235 - Section 4.1: Versions MUST be representable using a non-negative 32 bit integer.
If a BLF subscription exists for long enough, using %d may print negative version numbers.
Unlikely, as 2^32 at 1 update per second is ~137 years, or half that before the versions number started going negative.
Tested with Asterisk 1.8.8.2 with Grandstream phones.
alecdavis (license 585)
Tested by: alecdavis
Review: https://reviewboard.asterisk.org/r/1694/
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For whatever reason, we don't have a single function for copying data like this
from SIP peers to the SIP pvt. This patch adds the copying of amaflags to the
sip_pvt, but it would probably be worth discussing this function along with
the others that essentially just copy some amount of data from a peer to a
private.
(Closes issue ASTERISK-19029)
Reported by: Matt Lehner
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main/test.c includes "asterisk/version.h", when it should include
"asterisk/ast_version.h" instead (and it should use the ast_get_version()
and ast_get_version_num() functions). This commit modifies it to extract
the Asterisk version information using the proper APIs, and as a result means
that main/test.c no longer needs to be rebuilt when a Subversion checkout
is updated or modified.
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* Fix authenticate MESSAGE losing custom headers added by the MESSAGE_DATA
function in the authorization attempt.
* Pass up better From header contents for SIP to use. Now is in the
"display-name" <URI> format expected by MessageSend. (Note that this is a
behavior change that could concievably affect some people.)
* Block user from adding standard headers that are added automatically.
(To, From,...)
* Allow the user to override the Content-Type header contents sent by
MessageSend.
* Decrement Max-Forwards header if the user transferred it from an
incoming message.
* Expand SIP short header names so the dialplan and other code only has to
deal with the full names.
* Documents what SIP expects in the MessageSend(from) parameter.
(closes issue ASTERISK-18992)
Reported by: Yuri
(closes issue ASTERISK-18917)
Reported by: Shaun Clark
Review: https://reviewboard.asterisk.org/r/1683/
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These files have no need to include "asterisk/version.h", and doing so forces
them to be rebuilt each time a Subversion checkout moves between 'modified'
and 'unmodified' states.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@352516 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There was faulty information in the sample config describing user as a synonym for friend
so it has been changed to better elaborate on the differences between the three entity
types.
(closes issue ASTERISK-15537)
Reported by: yarique
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Now that we have the right license for the Russian 1.4.22 sounds as well as the sounds
for the Australian English 1.4.22 sounds, we can finally set the sounds to use 1.4.22!
(closes issue ASTERISK-18978)
Reported by: Cameron Twomey
Patches:
confbridge.tar.001 uploaded by Cameron Twomey
confbridge.tar.002 uploaded by Cameron Twomey
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While the FAXOPT function could be used to set the modem capabilities, the
input to that function was not being applied correctly to the spandsp layer.
This patch applies the current model capabilities before starting the spandsp
layer.
(closes issue: ASTERISK-16409)
Reported by: Kristijan Vrban
Tested by: Matt Jordan, Matthew Nicholson
Patches:
spandsp-modems-1.8.diff uploaded by mnicholson (license 5081)
spandsp-modems-10.diff uploaded by mnicholson (license 5081)
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Note: Noone calls ast_app_dtget() with the timeout parameter of zero so
the bad code normally will never get executed.
* Fix unnecessary floating point division in func_timeout.c
timeout_write() when all other values are integers.
(closes issue ASTERISK-16817)
Reported by: Dmitry Andrianov
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If a blind transfer were initiated using a REFER without a prior
reINVITE to place the call on hold, AND if Asterisk were sending
RTCP reports, then there was a reference for the RTP instance
of the transferer.
This fixes the issue by merging two similar but slightly conflicting
sections of code into a single area. It also adds a stop_media_flows()
call in the case that the transferer's UA never sends a BYE to us
like it is supposed to.
(issue ASTERISK-19192)
Review: https://reviewboard.asterisk.org/r/1681/
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