https://origsvn.digium.com/svn/asterisk/branches/1.8
(closes issue #18693)
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r310734 | may | 2011-03-15 00:45:53 +0300 (Tue, 15 Mar 2011) | 12 lines
Introduce t.38 parameters control functionality not full but enough for
Send/RcvFax support
Introduce t.38 controls between asterisk core and channel/proto layers.
Not all parameters are transferred from proto layers but *Fax apps
tested and work ok.
(issue #18693)
Reported by: benngard2
Patches:
issue-18693.patch uploaded by may213 (license 454)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310735 65c4cc65-6c06-0410-ace0-fbb531ad65f3
small fixes.
Interpret remote side H.225 version.
Corrections for H.323v2 endpoints:
don't start TCS and MSD before connect,
don't start TCS and MSD by accepting H.245 connection,
start TCS and MSD by StartH245 facility message.
Other fixes:
fix non zeroended remoteDisplayName issue, small fixes in call clearing
by closing H.245 connection, tcp keepalive introduced on TCP
connections (now is hardcoded, will be configurable in the future),
don't force H.245tunneling if FastStart is active, don't send Alerting
singal more than once per call.
(closes issue #18542)
Reported by: vmikhelson
Patches:
issue18542-final-3.patch uploaded by may213 (license 454)
Tested by: vmikhelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307396 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch is the foundation of an entire new way of looking at media in Asterisk.
The code present in this patch is everything required to complete phase1 of my
Media Architecture proposal. For more information about this project visit the link below.
https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal
The primary function of this patch is to convert all the usages of format
bitfields in Asterisk to use the new format and format_cap APIs. Functionally
no change in behavior should be present in this patch. Thanks to twilson
and russell for all the time they spent reviewing these changes.
Review: https://reviewboard.asterisk.org/r/1083/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Change order of sending Terminal Capability Set and MasterSlave
Determination packets, MSD send when TCS exchange procedure is done
(we send tcs ack to remote and we have remote tcs ack already
or we receive tcs ack from remote and we have send our tcs ack to
remote already). Some endpoints can work in this sequence only,
i suggest they can't work with both (tcs and msd) exchange procedures
simultaneously.
Also changed StartH245 facility message sending. It send on
incoming calls only due to some endpoints can't proccess properly
this facility messages on their incoming calls.
(closes issue #18433)
Reported by: MrHanMan
Patches:
tcs-msd-h245-3.patch uploaded by may213 (license 454)
Tested by: MrHanMan, may213
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299715 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Don't use GK ID if it's not presented in GK replies
Extract GK ID not only in GK confirm but in GK register confirm also
(closes issue #18401)
Reported by: MrHanMan
Patches:
no-gkid-2.patch uploaded by may213 (license 454)
Tested by: may213, MrHanMan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@298100 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r277568 | tilghman | 2010-07-16 16:54:29 -0500 (Fri, 16 Jul 2010) | 8 lines
Since we split values at the semicolon, we should store values with a semicolon as an encoded value.
(closes issue #17369)
Reported by: gkservice
Patches:
20100625__issue17369.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277773 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The purpose of this patch is to eliminate struct ast_callerid since it has
turned into a miscellaneous collection of various party information.
Eliminate struct ast_callerid and replace it with the following struct
organization:
struct ast_party_name {
char *str;
int char_set;
int presentation;
unsigned char valid;
};
struct ast_party_number {
char *str;
int plan;
int presentation;
unsigned char valid;
};
struct ast_party_subaddress {
char *str;
int type;
unsigned char odd_even_indicator;
unsigned char valid;
};
struct ast_party_id {
struct ast_party_name name;
struct ast_party_number number;
struct ast_party_subaddress subaddress;
char *tag;
};
struct ast_party_dialed {
struct {
char *str;
int plan;
} number;
struct ast_party_subaddress subaddress;
int transit_network_select;
};
struct ast_party_caller {
struct ast_party_id id;
char *ani;
int ani2;
};
The new organization adds some new information as well.
* The party name and number now have their own presentation value that can
be manipulated independently. ISDN supplies the presentation value for
the name and number at different times with the possibility that they
could be different.
* The party name and number now have a valid flag. Before this change the
name or number string could be empty if the presentation were restricted.
Most channel drivers assume that the name or number is then simply not
available instead of indicating that the name or number was restricted.
* The party name now has a character set value. SIP and Q.SIG have the
ability to indicate what character set a name string is using so it could
be presented properly.
* The dialed party now has a numbering plan value that could be useful to
have available.
The various channel drivers will need to be updated to support the new
core features as needed. They have simply been converted to supply
current functionality at this time.
The following items of note were either corrected or enhanced:
* The CONNECTEDLINE() and REDIRECTING() dialplan functions were
consolidated into func_callerid.c to share party id handling code.
* CALLERPRES() is now deprecated because the name and number have their
own presentation values.
* Fixed app_alarmreceiver.c write_metadata(). The workstring[] could
contain garbage. It also can only contain the caller id number so using
ast_callerid_parse() on it is silly. There was also a typo in the
CALLERNAME if test.
* Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id
number string. ast_callerid_parse() alters the given buffer which in this
case is the channel's caller id number string. Then using
ast_shrink_phone_number() could alter it even more.
* Fixed caller ID name and number memory leak in chan_usbradio.c.
* Fixed uninitialized char arrays cid_num[] and cid_name[] in
sig_analog.c.
* Protected access to a caller channel with lock in chan_sip.c.
* Clarified intent of code in app_meetme.c sla_ring_station() and
dial_trunk(). Also made save all caller ID data instead of just the name
and number strings.
* Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge()
function.
* Corrected some weirdness with app_privacy.c's use of caller
presentation.
Review: https://reviewboard.asterisk.org/r/702/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds a generic API for accommodating IPv6 and IPv4 addresses
within Asterisk. While many files have been updated to make use of the
API, chan_sip and the RTP code are the files which actually support
IPv6 addresses at the time of this commit. The way has been paved for
easier upgrading for other files in the near future, though.
Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne
for their hard work on this.
(closes issue #17565)
Reported by: russell
Patches:
asteriskv6-test-report.pdf uploaded by russell (license 2)
Review: https://reviewboard.asterisk.org/r/743
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274783 65c4cc65-6c06-0410-ace0-fbb531ad65f3
People expressed an interest in having access to the exact length of calls to a finer degree than seconds. See the CHANGES and UPGRADE.txt for usage also updated the sample configs to note the change.
Patch by snuffy.
(closes issue #16559)
Reported by: cianmaher
Tested by: cianmaher, snuffy
Review: https://reviewboard.asterisk.org/r/461/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269153 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Now that these files are in the tree, they should prefer the tree's local
copy of all Asterisk headers over any that may be installed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254931 65c4cc65-6c06-0410-ace0-fbb531ad65f3
added response to roundtrip delay requests from opposite side
added roundtrip delay request sending to opposite side after answer,
added options for sending request (interval between request and
count of unreplied requests before forced call hangup)
(closes issue #16976)
Reported by: vmikhelson
Patches:
rtdr-1.6.0-2.patch uploaded by may213 (license 454)
Tested by: vmikhelson, may213
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252277 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change basically reverts the change reviewed in
https://reviewboard.asterisk.org/r/374/ and instead limits the
updating of the RTP synchronization source to only those times when we
detect that the other side of the conversation has changed the ssrc.
The problem is that SRCUPDATE control frames are sent many times where
we don't want a new ssrc, including whenever Asterisk has to send DTMF
in a normal bridge. This is also not the first time that this mistake
has been made. The initial implementation of the ast_rtp_new_source
function also changed the ssrc--and then it was removed because of
this same issue. Then, we put it back in again to fix a different
issue. This patch attempts to only change the ssrc when we see that
the other side of the conversation has changed the ssrc.
It also renames some functions to make their purpose more clear.
Review: https://reviewboard.asterisk.org/r/540/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252089 65c4cc65-6c06-0410-ace0-fbb531ad65f3
incorrect q.931 message order filtered on incoming calls (first msg must be setup,
next must be not setup)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242645 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fixes a crash on Solaris.
(closes issue #16572)
Reported by: crjw
Patches:
frame_changes.patch uploaded by crjw (license 963)
Plus several others found and fixed by me
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239074 65c4cc65-6c06-0410-ace0-fbb531ad65f3
when we decode received q931 packet we must do callbacks and
when we print sended q931 packet we must not.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239037 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There is a bug when using ast_seekstream/ast_tellstream with format_mp3 in that
the file read position is not reset before attempting to read samples. So when
we seek to determine the maximum size of the file (as in res_agi's STREAM FILE)
we weren't then resetting the file pointer so that we could properly read
samples. This patch addresses that (in a similar manner to format_wav.c).
(closes issue #15224)
Reported by: rbd
Patches:
20091230_addons_1.4_issue15224.diff uploaded by seanbright (license 71)
Tested by: rbd, seanbright
Review: https://reviewboard.asterisk.org/r/453
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@238014 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In the process of swapping ULAW to a place in the extended codec space, we
found several unhandled cases, where a 32-bit integer was still being used to
handle a codec field. Most of these have been fixed with this commit, although
there is at least one case (codec_dahdi) which depends upon outside headers to
be altered before a conversion can be made.
(Fixes AST-278, SWP-459)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231850 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* chan_console accessed pvts after deallocation.
* cdr_mysql stored a pointer that was freed by realloc()
* The module loader did not check usecount on shutdown, which led to chan_iax2
reading a timer that was already unloaded.
* The event subsystem sometimes creates an event with no IEs. Due to a corner
condition, the code would read beyond the memory boundary.
* res_pktccops did not correctly check whether its monitor thread was started.
(closes issue #16062)
Reported by: alexanderheinz
Patches:
20091109__issue16062.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
All addons modules should be disabled by default, requiring the
user to turn them on if desired. After all, these are addons we're
talking about here.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228659 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For some reason, the code compiles just fine with later versions
of GCC, but this one requires some weird double casting in order
to get rid of all warnings. Whatever.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228658 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Many architectural and functional changes.
Main changes are threading model chanes (many thread in ooh323 stack
instead of one), modifications and improvements in signalling part,
additional codecs support (726, speex), t38 mode support.
This module tested and used in production environment.
(closes issue #15285)
Reported by: may213
Tested by: sles, c0w, OrNix
Review: https://reviewboard.asterisk.org/r/324/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227898 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This problem was introduced when the AST_FRIENDLY_OFFSET patch was merged.
I'm surprised that nobody noticed any trouble when testing that patch, but this
fixes the code that fills in the buffer to start filling in after the offset
portion of the buffer.
(closes issue #15850)
Reported by: 99gixxer
Patches:
issue15850.diff1.txt uploaded by russell (license 2)
Tested by: 99gixxer
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217113 65c4cc65-6c06-0410-ace0-fbb531ad65f3
format_mp3 claimed that it provided AST_FRIENDLY_OFFSET in frames returned by
read(). However, it lied. This means that other parts of the code that
attempted to make use of the offset buffer would end up corrupting the fields
in the ast_filestream structure. This resulted in quite a few crashes due to
unexpected values for fields in ast_filestream.
This patch closes out quite a few bugs. However, some of these bugs have been
open for a while and have been an area where more than one bug has been
discussed. So with that said, anyone that is following one of the issues
closed here, if you still have a problem, please open a new bug report for the
specific problem you are still having. If you do, please ensure that the bug
report is based on the newest version of Asterisk, and that this patch is
applied if format_mp3 is in use. Thanks!
(closes issue #15109)
Reported by: jvandal
Tested by: aragon, russell, zerohalo, marhbere, rgj
(closes issue #14958)
Reported by: aragon
(closes issue #15123)
Reported by: axisinternet
(closes issue #15041)
Reported by: maxnuv
(closes issue #15396)
Reported by: aragon
(closes issue #15195)
Reported by: amorsen
Tested by: amorsen
(closes issue #15781)
Reported by: jensvb
(closes issue #15735)
Reported by: thom4fun
(closes issue #15460)
Reported by: marhbere
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r207647 | kpfleming | 2009-07-21 08:04:44 -0500 (Tue, 21 Jul 2009) | 12 lines
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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207680 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Ensure that add-on modules can be embedded, fix up Makefile.moddir_rules
to allow module directory Makefiles to more easily specify the modules to
be built, and explicitly list the addons modules in its Makefile, since
the module names don't follow any pattern.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204415 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Someone asked yesterday, "is there a good reason why we can't just put these
modules in Asterisk?". After a brief discussion, as long as the modules are
clearly set aside in their own directory and not enabled by default, it is
perfectly fine.
For more information about why a module goes in addons, see README-addons.txt.
chan_ooh323 does not currently compile as it is behind some trunk API updates.
However, it will not build by default, so it should be okay for now.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Alexandr Anikin
Paul Belanger
Tilghman Lesher
David Vossel
Russell Bryant
Matthew Nicholson
Erin Spiceland
Jason Parker
Mark Michelson
Richard Mudgett
Tzafrir Cohen
Bradley Latus
Kevin P. Fleming
Terry Wilson
Sean Bright
Olle Johansson
Jeff Peeler
David Brooks