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See RFC 5545 ch. 3.8.1.2 and 9.
(closes issue #17837)
Review: https://reviewboard.asterisk.org/r/880/
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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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r284477 | twilson | 2010-09-01 13:44:36 -0500 (Wed, 01 Sep 2010) | 17 lines
Fix SRTP for changing SSRC and multiple a=crypto SDP lines
Adding code to Asterisk that changed the SSRC during bridges and masquerades
broke SRTP functionality. Also broken was handling the situation where an
incoming INVITE had more than one crypto offer. This patch caches the SRTP
policies the we use so that we can change the ssrc and inform libsrtp of the
new streams. It also uses the first acceptable a=crypto line from the incoming
INVITE.
(closes issue #17563)
Reported by: Alexcr
Patches:
srtp.diff uploaded by twilson (license 396)
Tested by: twilson
Review: https://reviewboard.asterisk.org/r/878/
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r284473 | tilghman | 2010-09-01 13:16:37 -0500 (Wed, 01 Sep 2010) | 12 lines
Merged revisions 284472 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r284472 | tilghman | 2010-09-01 13:13:35 -0500 (Wed, 01 Sep 2010) | 5 lines
Don't warn on floats and timestamps
(closes issue #17082)
Reported by: coolmig
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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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r283770 | tilghman | 2010-08-26 18:47:02 -0500 (Thu, 26 Aug 2010) | 8 lines
Convert MOH to use generic timers.
(closes issue #17726)
Reported by: lmadsen
Patches:
20100825__issue17726__2.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
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r283659 | russell | 2010-08-26 08:26:14 -0500 (Thu, 26 Aug 2010) | 2 lines
Slight improvement to a debug message.
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r283561 | tilghman | 2010-08-25 11:12:43 -0500 (Wed, 25 Aug 2010) | 5 lines
Initialize connect timeout on each time through the loop.
(closes issue #17911)
Reported by: wurstsalat
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r283457 | lmadsen | 2010-08-24 13:56:29 -0500 (Tue, 24 Aug 2010) | 9 lines
Fix issue where TOS is no longer set on RTP packets.
Fix issue where the tos is no longer being set on RTP packets through res_rtp_asterisk.
(closes issue #17890)
Reported by: elguero
Patches:
qos_18.diff uploaded by elguero (license 37)
Review: https://reviewboard.asterisk.org/r/868
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a WARNING message.
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r283175 | tilghman | 2010-08-23 07:06:26 -0500 (Mon, 23 Aug 2010) | 2 lines
Don't fail to start if the config file is missing.
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r282269 | dvossel | 2010-08-13 15:03:56 -0500 (Fri, 13 Aug 2010) | 4 lines
res_stun_monitor for monitoring network changes behind a NAT device
Review: https://reviewboard.asterisk.org/r/854
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r281358 | mnicholson | 2010-08-09 09:49:38 -0500 (Mon, 09 Aug 2010) | 4 lines
Validate minrate, maxrate, and modem settings before attempting a fax session.
FAX-224
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r280909 | mnicholson | 2010-08-04 10:11:13 -0500 (Wed, 04 Aug 2010) | 2 lines
Initialize FAXOPT() status variables in sendfax and receivefax instead of when the details structure is created.
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r280557 | mnicholson | 2010-07-29 16:07:21 -0500 (Thu, 29 Jul 2010) | 4 lines
Fix regression introduced in r1664. Give the fax stack time to shutdown and populate the FAXOPT output variables.
FAX-222
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r280225 | twilson | 2010-07-28 12:34:42 -0700 (Wed, 28 Jul 2010) | 3 lines
Do rtp/rtcp debugging when it is turned on w/o filtering
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r280058 | russell | 2010-07-28 08:01:15 -0500 (Wed, 28 Jul 2010) | 2 lines
s/init keys/keys init/
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r279601 | tilghman | 2010-07-26 16:07:45 -0500 (Mon, 26 Jul 2010) | 19 lines
Merged revisions 279597 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r279597 | ghenry | 2010-07-26 15:25:54 -0500 (Mon, 26 Jul 2010) | 13 lines
Apply all patches in:
https://issues.asterisk.org/view.php?id=13573
(closes issue #13573)
Reported by: navkumar
Patches:
res_config_ldap-category.diff uploaded by navkumar (license 580)
res_config_ldap.patch uploaded by bencer (license 961)
res_config_ldap uploaded by bencer (license 961)
Tested by: suretec
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r279113 | tilghman | 2010-07-23 13:56:04 -0500 (Fri, 23 Jul 2010) | 2 lines
Silly 64-bit compilers (who uses 64-bit anyway?)
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Review: https://reviewboard.asterisk.org/r/798/
Thanks Mark for a quick review!
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Review: https://reviewboard.asterisk.org/r/793/
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This code did not properly check FD_SETSIZE to ensure that it did not try to
select() on fds that were too large. Switching to poll() removes the limitation
on the maximum fd value.
(closes issue #15915)
Reported by: keiron
(closes issue #17187)
Reported by: Eddie Edwards
(closes issue #16494)
Reported by: Hubguru
(closes issue #15731)
Reported by: flop
(closes issue #12917)
Reported by: falves11
(closes issue #14920)
Reported by: vrban
(closes issue #17199)
Reported by: aleksey2000
(closes issue #15406)
Reported by: kowalma
(closes issue #17438)
Reported by: dcabot
(closes issue #17325)
Reported by: glwgoes
(closes issue #17118)
Reported by: erikje
possibly other issues, too ...
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actual failure.
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to be used, even when realtime is used.
(closes issue #17082)
Reported by: coolmig
Patches:
20100720__issue17082.diff.txt uploaded by tilghman (license 14)
Tested by: coolmig
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handled.
FAX output channel variables will now match the values reported by FAXOPT() and should be set in all failure and success cases.
This commit also contains a few modifications to the way FAXOPT() variables are populated in a few spots and fixes for some reference count leaks of the session details structure in some failure cases.
Also found and fixed more cases where FAXOPT(status) may not have gotten set.
FAX-214
FAX-203
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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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(closes issues #17667)
Reported by: snuffy
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ast_sockaddr_stringiy_fmt (which is call by all ast_sockaddr_stringify* functions)
uses thread-local storage for storing the string that it creates. In cases where
ast_sockaddr_stringify_fmt was being called twice within the same statement, the
result of one call would be overwritten by the result of the other call. This
usually was happening in printf-like statements and was resulting in the same
stringified addressed being printed twice instead of two separate addresses.
I have fixed this by using ast_strdupa on the result of stringify functions if
they are used twice within the same statement. As far as I could tell, there were
no instances where a pointer to the result of such a call were saved anywhere, so
this is the only situation I could see where this error could occur.
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(closes issue #17475)
Reported by: tilghman
Review: https://reviewboard.asterisk.org/r/695/
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Expand the ani field in ast_party_caller and ast_party_connected_line to
an ast_party_id.
This is an extension to the ast_callerid restructuring patch in review:
https://reviewboard.asterisk.org/r/702/
Review: https://reviewboard.asterisk.org/r/744/
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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/
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tracking down the source.
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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
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midterm evaluation.
Review: https://reviewboard.asterisk.org/r/757/
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possible. Previously some failure cases did not result in proper FAXOPT values.
FAX-203
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dialplan function. Also make fax_rate_str_to_int() return an unsigned int and return 0 instead of -1 in the event of an error.
FAX-202
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r274157 | mmichelson | 2010-07-06 09:29:23 -0500 (Tue, 06 Jul 2010) | 16 lines
Fix problem with RFC 2833 DTMF not being accepted.
A recent check was added to ensure that we did not erroneously
detect duplicate DTMF when we received packets out of order.
The problem was that the check did not account for the fact that
the seqno of an RTP stream will roll over back to 0 after hitting
65535. Now, we have a secondary check that will ensure that the
seqno rolling over will not cause us to stop accepting DTMF.
(closes issue #17571)
Reported by: mdeneen
Patches:
rtp_seqno_rollover.patch uploaded by mmichelson (license 60)
Tested by: richardf, maxochoa, JJCinAZ
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(Also fix the typos in the comments)
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FAX-177
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Also clean up some coding errors.
(closes issue #17469)
Reported by: wdoekes
Patches:
astsvn-rtp-set-debug-ip.patch uploaded by wdoekes (license 717)
Tested by: wdoekes, pabelanger
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the config file also changes.
(closes issue #16982)
Reported by: dmitri
Patches:
res_musiconhold.patch uploaded by dmitri (license 1001)
Tested by: atis
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272332 f38db490-d61c-443f-a65b-d21fe96a405b
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