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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r292122 | mnicholson | 2010-10-18 12:15:24 -0500 (Mon, 18 Oct 2010) | 5 lines
Fix the cmgr parser.
(closes issue 0018152)
Reported by: menschentier
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@292124 f38db490-d61c-443f-a65b-d21fe96a405b
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Added options for faststart/h.245 tunneling per user/peer, properly
handle these and global options, correction of handling fs/tunneling
fields in signalling responses
(closes issue #17972)
Reported by: salecha
Patches:
fs-tunnel-per-point-3.patch uploaded by may213 (license 454)
Tested by: may213, salecha
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@291006 f38db490-d61c-443f-a65b-d21fe96a405b
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MySQL should use.
(closes issue 17948)
Reported by qmax.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@290939 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@288893 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r285336 | tilghman | 2010-09-07 14:38:12 -0500 (Tue, 07 Sep 2010) | 2 lines
Fix build on FreeBSD 8.0, take 2.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@285337 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r284696 | tilghman | 2010-09-02 11:27:52 -0500 (Thu, 02 Sep 2010) | 2 lines
Fixing build
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@284700 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r284597 | tilghman | 2010-09-02 00:00:34 -0500 (Thu, 02 Sep 2010) | 29 lines
Merged revisions 284593,284595 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r284593 | tilghman | 2010-09-01 17:59:50 -0500 (Wed, 01 Sep 2010) | 18 lines
Merged revisions 284478 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r284478 | tilghman | 2010-09-01 13:49:11 -0500 (Wed, 01 Sep 2010) | 11 lines
Ensure that all areas that previously used select(2) now use poll(2), with implementations that need poll(2) implemented with select(2) safe against 1024-bit overflows.
This is a followup to the fix for the pthread timer in 1.6.2 and beyond, fixing
a potential crash bug in all supported releases.
(closes issue #17678)
Reported by: russell
Branch: https://origsvn.digium.com/svn/asterisk/team/tilghman/ast_select
Review: https://reviewboard.asterisk.org/r/824/
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r284595 | tilghman | 2010-09-01 22:57:43 -0500 (Wed, 01 Sep 2010) | 2 lines
Failed to rerun bootstrap.sh after last commit
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@284598 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r283882 | qwell | 2010-08-27 15:31:55 -0500 (Fri, 27 Aug 2010) | 22 lines
Merged revisions 283881 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r283881 | qwell | 2010-08-27 15:30:27 -0500 (Fri, 27 Aug 2010) | 15 lines
Merged revisions 283880 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r283880 | qwell | 2010-08-27 15:29:11 -0500 (Fri, 27 Aug 2010) | 8 lines
Fix issue with decoding ^-escaped characters in realtime.
(closes issue #17790)
Reported by: denzs
Patches:
17790-chunky.diff uploaded by qwell (license 4)
Tested by: qwell, denzs
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@283883 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r280742 | russell | 2010-08-03 13:48:45 -0500 (Tue, 03 Aug 2010) | 4 lines
Remove the MP3 decoder source code and replace it with a small shell script.
Review: https://reviewboard.asterisk.org/r/836/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@280743 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278943 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278132 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@277773 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274827 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@274783 f38db490-d61c-443f-a65b-d21fe96a405b
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(Also fix the typos in the comments)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@273641 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16430)
Reported by: azbest
Tested by: azbest
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@273312 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@269153 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@265525 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@265227 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261964 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258855 f38db490-d61c-443f-a65b-d21fe96a405b
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Don't pass zero callerid string to ooh323 stack because it can't encode this properly and
can't generate setup message.
(closes issue #17186)
Reported by: vmikhelson
Patches:
zero_callerid_num.patch uploaded by may213 (license 454)
Tested by: may213
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258838 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@255199 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@254931 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@254884 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@254801 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@254799 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@254636 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@252277 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@252089 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@251181 f38db490-d61c-443f-a65b-d21fe96a405b
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Tested by: benngard
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@247035 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@242933 f38db490-d61c-443f-a65b-d21fe96a405b
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incorrect q.931 message order filtered on incoming calls (first msg must be setup,
next must be not setup)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@242645 f38db490-d61c-443f-a65b-d21fe96a405b
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(typically, UTC)
(closes issue #16401)
Reported by: lordmortis
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@239152 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@239074 f38db490-d61c-443f-a65b-d21fe96a405b
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when we decode received q931 packet we must do callbacks and
when we print sended q931 packet we must not.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@239037 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@238014 f38db490-d61c-443f-a65b-d21fe96a405b
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don't decode UUIE from Q931StatusMessage
clean call without callIdentifier data
don't start tcs/msd exchange procedure after call proceeding received
(closes issue #16365)
Reported by: benngard2
Tested by: may213, benngard2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@237098 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@233732 f38db490-d61c-443f-a65b-d21fe96a405b
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the realtime API to work correctly.
(closes issue #16138)
Reported by: sohosys
Patches:
20091029__issue16138.diff.txt uploaded by tilghman (license 14)
Tested by: sohosys
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@233050 f38db490-d61c-443f-a65b-d21fe96a405b
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correction of double pointer references from previous rev
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@232853 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16278)
Reported by: Artem
Patches:
multiline-sms-fix2.diff uploaded by mnicholson (license 96)
Tested by: Artem
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@232580 f38db490-d61c-443f-a65b-d21fe96a405b
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use a chan_mobile endpoint that does not have service.
(closes issue #16132)
Reported by: nikkk
Patches:
service-indicator2.diff uploaded by mnicholson (license 96)
Tested by: nikkk
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@232544 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@231850 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@229840 f38db490-d61c-443f-a65b-d21fe96a405b
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* 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: http://svn.digium.com/svn/asterisk/trunk@228798 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@228659 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@228658 f38db490-d61c-443f-a65b-d21fe96a405b
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