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r285931 | tilghman | 2010-09-09 20:25:50 -0500 (Thu, 09 Sep 2010) | 21 lines
Merged revisions 285930 via svnmerge from
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r285930 | tilghman | 2010-09-09 20:16:32 -0500 (Thu, 09 Sep 2010) | 14 lines
Merged revisions 285889 via svnmerge from
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r285889 | tilghman | 2010-09-09 19:13:45 -0500 (Thu, 09 Sep 2010) | 7 lines
Fix Mac OS X build.
This also fixes a rather grievous calculation error for the offset of
ast_fdset, which was masked on Linux and FreeBSD, because these platforms
check the first 256 FDs regardless of the bitmask setting (due to backwards
compatibility).
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r285819 | pabelanger | 2010-09-09 18:52:31 -0400 (Thu, 09 Sep 2010) | 22 lines
Merged revisions 285818 via svnmerge from
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r285818 | pabelanger | 2010-09-09 18:49:19 -0400 (Thu, 09 Sep 2010) | 15 lines
Merged revisions 285817 via svnmerge from
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r285817 | pabelanger | 2010-09-09 18:34:35 -0400 (Thu, 09 Sep 2010) | 8 lines
GCC 4.2.x optimizations result in improper behavior of GSM codec
(closes issue #17688)
Reported by: pprindeville
Patches:
asterisk-trunk-bugid11243.patch uploaded by pprindeville (license 347)
Tested by: mkeuter, pprindeville
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r285745 | qwell | 2010-09-09 15:11:06 -0500 (Thu, 09 Sep 2010) | 23 lines
Merged revisions 285744 via svnmerge from
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r285744 | qwell | 2010-09-09 15:09:23 -0500 (Thu, 09 Sep 2010) | 16 lines
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r285742 | qwell | 2010-09-09 15:06:31 -0500 (Thu, 09 Sep 2010) | 9 lines
Transmit silence when reading DTMF in ast_readstring.
Otherwise, you could get issues with DTMF timeouts causing hangups.
(closes issue #17370)
Reported by: makoto
Patches:
channel-readstring-silence-generator.patch uploaded by makoto (license 38)
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r285711 | bbryant | 2010-09-09 14:51:52 -0400 (Thu, 09 Sep 2010) | 15 lines
Merged revisions 285710 via svnmerge from
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r285710 | bbryant | 2010-09-09 14:50:13 -0400 (Thu, 09 Sep 2010) | 8 lines
Fixes an issue with dialplan pattern matching where the specificity for pattern ranges and pattern special characters was inconsistent.
(closes issue #16903)
Reported by: Nick_Lewis
Patches:
pbx.c-specificity.patch uploaded by Nick Lewis (license 657)
Tested by: Nick_Lewis
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r285640 | bbryant | 2010-09-09 13:23:28 -0400 (Thu, 09 Sep 2010) | 21 lines
Merged revisions 285639 via svnmerge from
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r285639 | bbryant | 2010-09-09 13:22:25 -0400 (Thu, 09 Sep 2010) | 14 lines
Merged revisions 285638 via svnmerge from
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r285638 | bbryant | 2010-09-09 13:20:17 -0400 (Thu, 09 Sep 2010) | 7 lines
Fixes an issue with MOH where it doesn't recover cleanly when it can't play a file and would just stop, instead of continuing to find the next playable file in the MOH class.
(closes issue #17807)
Reported by: kshumard
Review: https://reviewboard.asterisk.org/r/910/
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r285568 | dvossel | 2010-09-08 17:14:19 -0500 (Wed, 08 Sep 2010) | 16 lines
Merged revisions 285567 via svnmerge from
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r285567 | dvossel | 2010-09-08 17:11:28 -0500 (Wed, 08 Sep 2010) | 9 lines
Merged revisions 285566 via svnmerge from
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r285566 | dvossel | 2010-09-08 17:07:31 -0500 (Wed, 08 Sep 2010) | 2 lines
In retrans_pkt, do not unlock pvt until the end of the function on a transmit failure.
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r285564 | dvossel | 2010-09-08 16:48:37 -0500 (Wed, 08 Sep 2010) | 60 lines
Merged revisions 285563 via svnmerge from
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r285563 | dvossel | 2010-09-08 16:47:29 -0500 (Wed, 08 Sep 2010) | 54 lines
Fixes interoperability problems with session timer behavior in Asterisk.
CHANGES:
1. Never put "timer" in "Require" header. This is not to our benefit
and RFC 4028 section 7.1 even warns against it. It is possible for one
endpoint to perform session-timer refreshes while the other endpoint does
not support them. If in this case the end point performing the refreshing
puts "timer" in the Require field during a refresh, the dialog will
likely get terminated by the other end.
2. Change the behavior of 'session-timer=accept' in sip.conf (which is
the default behavior of Asterisk with no session timer configuration
specified) to only run session-timers as result of an incoming INVITE
request if the INVITE contains an "Session-Expires" header... Asterisk is
currently treating having the "timer" option in the "Supported" header as
a request for session timers by the UAC. I do not agree with this. Session
timers should only be negotiated in "accept" mode when the incoming INVITE
supplies a "Session-Expires" header, otherwise RFC 4028 says we should
treat a request containing no "Session-Expires" header as a session with
no expiration.
Below I have outlined some situations and what Asterisk's behavior is.
The table reflects the behavior changes implemented by this patch.
SITUATIONS:
-Asterisk as UAS
1. Incoming INVITE: NO "Session-Expires"
2. Incoming INVITE: HAS "Session-Expires"
-Asterisk as UAC
3. Outgoing INVITE: NO "Session-Expires". 200 Ok Response HAS "Session-Expires" header
4. Outgoing INVITE: NO "Session-Expires". 200 Ok Response NO "Session-Expires" header
5. Outgoing INVITE: HAS "Session-Expires".
Active - Asterisk will have an active refresh timer regardless if the other endpoint does.
Inactive - Asterisk does not have an active refresh timer regardless if the other endpoint does.
XXXXXXX - Not possible for mode.
______________________________________
|SITUATIONS | 'session-timer' MODES |
|___________|________________________|
| | originate | accept |
|-----------|------------|-----------|
|1. | Active | Inactive |
|2. | Active | Active |
|3. | XXXXXXXX | Active |
|4. | XXXXXXXX | Inactive |
|5. | Active | XXXXXXXX |
--------------------------------------
(closes issue #17005)
Reported by: alexrecarey
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r285533 | bbryant | 2010-09-08 16:58:43 -0400 (Wed, 08 Sep 2010) | 15 lines
Merged revisions 285532 via svnmerge from
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r285532 | bbryant | 2010-09-08 16:56:12 -0400 (Wed, 08 Sep 2010) | 8 lines
Fixes a bug with MeetMe where after announcing the amount of time left in a conference, if music on hold was playing, it doesn't restart.
(closes issue #17408)
Reported by: sysreq
Patches:
asterisk-issue-17408_fixed.patch uploaded by sysreq (license 1009)
Tested by: sysreq
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r285530 | qwell | 2010-09-08 15:43:10 -0500 (Wed, 08 Sep 2010) | 9 lines
Merged revisions 285529 via svnmerge from
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r285529 | qwell | 2010-09-08 15:42:44 -0500 (Wed, 08 Sep 2010) | 1 line
Follow coding guidelines in moh rescan fix. Also fix the documentation that got me in trouble.
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r285527 | qwell | 2010-09-08 15:32:13 -0500 (Wed, 08 Sep 2010) | 15 lines
Merged revisions 285526 via svnmerge from
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r285526 | qwell | 2010-09-08 15:31:43 -0500 (Wed, 08 Sep 2010) | 8 lines
Fixes issue where moh files were no longer rescanned during a reload.
(closes issue #16744)
Reported by: pj
Patches:
16744-reload.diff uploaded by qwell (license 4)
Tested by: qwell
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r285484 | tilghman | 2010-09-08 02:14:17 -0500 (Wed, 08 Sep 2010) | 2 lines
Documentation only
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r285455 | qwell | 2010-09-07 17:22:14 -0500 (Tue, 07 Sep 2010) | 8 lines
Don't automatically add domains for wildcard bindaddrs.
(closes issue #17832)
Reported by: oej
Patches:
17832-wildcard.diff uploaded by qwell (license 4)
Tested by: qwell
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r285369 | qwell | 2010-09-07 15:58:34 -0500 (Tue, 07 Sep 2010) | 7 lines
Add note to 'sip show settings' regarding dual-stack support, and a :: bindaddress.
(closes issue #17831)
Reported by: oej
Patches:
17831-v6wildcardbind.diff uploaded by qwell (license 4)
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r285386 | tilghman | 2010-09-07 16:20:16 -0500 (Tue, 07 Sep 2010) | 13 lines
Don't notify on attribute changes, and change how the queuing mechanism works.
Fixes call spools in 1.8.
(closes issue #17337)
Reported by: loloski
Patches:
20100827__issue17337.diff.txt uploaded by tilghman (license 14)
(closes issue #17924)
Reported by: mkeuter
Tested by: mkeuter
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r285373 | tilghman | 2010-09-07 16:14:03 -0500 (Tue, 07 Sep 2010) | 7 lines
Add CHANNEL(checkhangup) to check whether a channel is in the process of being hanged up.
(closes issue #17652)
Reported by: kobaz
Patches:
func_channel.patch uploaded by kobaz (license 834)
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r285371 | rmudgett | 2010-09-07 16:08:35 -0500 (Tue, 07 Sep 2010) | 1 line
Fix cut-n-paste error.
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r285367 | tilghman | 2010-09-07 15:56:07 -0500 (Tue, 07 Sep 2010) | 23 lines
Merged revisions 285366 via svnmerge from
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r285366 | tilghman | 2010-09-07 15:31:41 -0500 (Tue, 07 Sep 2010) | 16 lines
Merged revisions 285365 via svnmerge from
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r285365 | tilghman | 2010-09-07 15:30:22 -0500 (Tue, 07 Sep 2010) | 9 lines
Catch invalid extensions at the parser, instead of making the core deal with them.
(closes issue #17794)
Reported by: PavelL
Patches:
20100820__issue17794__1.6.2.diff.txt uploaded by tilghman (license 14)
20100820__issue17794__1.4.diff.txt uploaded by tilghman (license 14)
Tested by: PavelL
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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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r285268 | tilghman | 2010-09-07 14:08:09 -0500 (Tue, 07 Sep 2010) | 18 lines
Merged revisions 285267 via svnmerge from
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r285267 | tilghman | 2010-09-07 14:07:17 -0500 (Tue, 07 Sep 2010) | 11 lines
Merged revisions 285266 via svnmerge from
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r285266 | tilghman | 2010-09-07 14:04:50 -0500 (Tue, 07 Sep 2010) | 4 lines
Use poll, if indicated to do so, in the ast_poll2 implementation.
This fixes the unit tests on FreeBSD 8.0.
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r285197 | bbryant | 2010-09-07 13:54:21 -0400 (Tue, 07 Sep 2010) | 24 lines
Merged revisions 285196 via svnmerge from
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r285196 | bbryant | 2010-09-07 13:49:07 -0400 (Tue, 07 Sep 2010) | 17 lines
Merged revisions 285194 via svnmerge from
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r285194 | bbryant | 2010-09-07 13:45:41 -0400 (Tue, 07 Sep 2010) | 10 lines
Fixes voicemail.conf issues where mailboxes with passwords that don't precede a comma would throw unnecessary error messages.
(closes issue #15726)
Reported by: 298
Patches:
M15726.diff uploaded by junky (license 177)
Tested by: junky
Review: [full review board URL with trailing slash]
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r285195 | rmudgett | 2010-09-07 12:47:34 -0500 (Tue, 07 Sep 2010) | 20 lines
Merged revisions 285193 via svnmerge from
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Merged revisions 285192 via svnmerge from
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r285192 | rmudgett | 2010-09-07 11:58:57 -0500 (Tue, 07 Sep 2010) | 8 lines
COLP/CONP and chan_misdn missing update
chan_misdn does not update the caller id of the channel if a new connected
number or ECT-INFORM (w/ new peer number on call transfer) is received.
JIRA ABE-2502
JIRA SWP-2058
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r285161 | russell | 2010-09-06 15:10:03 -0500 (Mon, 06 Sep 2010) | 4 lines
Fix libsrtp -fPIC check for when non-standard prefix is used.
Thanks to loompek in #asterisk for reporting the issue and testing this patch.
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r285162 | russell | 2010-09-06 15:10:24 -0500 (Mon, 06 Sep 2010) | 1 line
regenerate configure script.
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r285090 | tilghman | 2010-09-06 01:56:07 -0500 (Mon, 06 Sep 2010) | 16 lines
Merged revisions 285089 via svnmerge from
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r285089 | tilghman | 2010-09-06 01:55:17 -0500 (Mon, 06 Sep 2010) | 9 lines
Merged revisions 285088 via svnmerge from
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r285088 | tilghman | 2010-09-06 01:54:18 -0500 (Mon, 06 Sep 2010) | 2 lines
Silly convenience script for BSD platforms.
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r285057 | russell | 2010-09-04 13:08:19 -0500 (Sat, 04 Sep 2010) | 2 lines
Add a C++ compatible version of AST_CLI_DEFINE().
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r285017 | twilson | 2010-09-03 18:19:54 -0500 (Fri, 03 Sep 2010) | 4 lines
Call correct lock function as transferer is a sip_pvt not a channel
Both functions are #defined to ao2_lock, but still...
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r285006 | dvossel | 2010-09-03 17:21:50 -0500 (Fri, 03 Sep 2010) | 9 lines
Disables auth_options_request option by default.
The auth_options_request option was created to do authentication
on OPTIONS request just like INVITES are done. Since it has been
noted that some endpoints use OPTIONS requests as a way of qualifying
a peer and that a 401 authentication response could result in
interoperability issues, this option has been disabled by default.
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r284967 | bbryant | 2010-09-03 14:19:53 -0400 (Fri, 03 Sep 2010) | 15 lines
Merged revisions 284958 via svnmerge from
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r284958 | bbryant | 2010-09-03 14:15:49 -0400 (Fri, 03 Sep 2010) | 8 lines
This is a patch provided for issue #17935 to add the ActionID to the IAXregistry AMI response.
(closes issue #17935)
Reported by: alexkuklin
Patches:
iaxshowreg uploaded by alexkuklin (license 1115)
Tested by: alexkuklin
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r284952 | dvossel | 2010-09-03 13:03:23 -0500 (Fri, 03 Sep 2010) | 2 lines
During OPTIONS authentication, the authpeer does not need to be returned for any reason.
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r284950 | dvossel | 2010-09-03 12:29:02 -0500 (Fri, 03 Sep 2010) | 14 lines
authenticate OPTIONS requests just like we would an INVITE
OPTIONS requests should be treated the same as an INVITE
This includes authentication. This patch adds the ability for
incoming out of dialog OPTION requests to be authenticated
before providing a response indicating whether an extension
is available or not. The authentication routine works the
exact same way as it does for incoming INVITEs. This means
that if a peer has 'insecure=invite' in their peer definition,
the same will be true for the processing of the OPTIONS request.
Review: https://reviewboard.asterisk.org/r/881/
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r284921 | twilson | 2010-09-03 11:28:18 -0500 (Fri, 03 Sep 2010) | 19 lines
Merged revisions 284897 via svnmerge from
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r284897 | twilson | 2010-09-03 11:20:45 -0500 (Fri, 03 Sep 2010) | 12 lines
Merged revisions 284881 via svnmerge from
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r284881 | twilson | 2010-09-03 11:10:23 -0500 (Fri, 03 Sep 2010) | 5 lines
Properly detect when a sound file doesn't exist
ast_fileexists returns -1 for error and 0 for a non-existant file. The existing
code treated missing files as though they existed.
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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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r284779 | rmudgett | 2010-09-02 15:59:12 -0500 (Thu, 02 Sep 2010) | 8 lines
Made output libpri event names if pri debugging is enabled when sig_pri processes them.
* Simplified CLI "pri debug xx span xx" command code and removed redundant
debugging enabled messages.
* Made CLI "pri debug xx span xx" command only close the debugging log
file if it was opened.
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r284780 | rmudgett | 2010-09-02 16:02:54 -0500 (Thu, 02 Sep 2010) | 2 lines
Simplified pri_dchannel() poll timeout duration code.
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r284705 | dvossel | 2010-09-02 11:56:43 -0500 (Thu, 02 Sep 2010) | 20 lines
Merged revisions 284704 via svnmerge from
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r284704 | dvossel | 2010-09-02 11:48:51 -0500 (Thu, 02 Sep 2010) | 13 lines
Merged revisions 284703 via svnmerge from
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r284703 | dvossel | 2010-09-02 11:47:15 -0500 (Thu, 02 Sep 2010) | 7 lines
Removed relatedpeer code from sip_autodestruct
Handling of the relatedpeer structure associated with a
sip_pvt should be done during the final sip_destruction
function, not in sip_autodestruct.
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r284701 | qwell | 2010-09-02 11:43:09 -0500 (Thu, 02 Sep 2010) | 8 lines
Add slin16 support for format_wav (new wav16 file extension)
(closes issue #15029)
Reported by: andrew
Patches:
wav16.patch uploaded by andrew (license 240)
Tested by: qwell, andrew
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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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r284698 | rmudgett | 2010-09-02 11:34:32 -0500 (Thu, 02 Sep 2010) | 5 lines
Added documentation for CONNECTEDLINE and REDIRECTING functions.
(closes issue #17808)
Reported by: jtodd
Review: https://reviewboard.asterisk.org/r/875/
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r284666 | tilghman | 2010-09-02 11:11:15 -0500 (Thu, 02 Sep 2010) | 9 lines
Merged revisions 284665 via svnmerge from
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r284665 | tilghman | 2010-09-02 11:07:19 -0500 (Thu, 02 Sep 2010) | 2 lines
Fixing build.
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r284632 | tilghman | 2010-09-02 00:31:02 -0500 (Thu, 02 Sep 2010) | 14 lines
Merged revisions 284631 via svnmerge from
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r284631 | tilghman | 2010-09-02 00:30:16 -0500 (Thu, 02 Sep 2010) | 7 lines
Don't reset queue stats on a module reload.
(closes issue #17535)
Reported by: raarts
Patches:
20100819__issue17535.diff.txt uploaded by tilghman (license 14)
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r284610 | tilghman | 2010-09-02 00:20:59 -0500 (Thu, 02 Sep 2010) | 10 lines
When optional_api is non-optional, force dependent modules to be loaded.
(closes issue #17707)
Reported by: ira
Patches:
20100819__issue17707__asterisk1.8.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
Review: https://reviewboard.asterisk.org/r/876/
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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
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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
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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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r284561 | dvossel | 2010-09-01 16:47:01 -0500 (Wed, 01 Sep 2010) | 9 lines
During request to dialog matching, verify init_ruri is present before comparing.
During request to dialog matching, we attempt a best effort routine for fork
detection which requires several elements to be in place. The dialog's
initial request uri is one of those elements. Since it is best effort,
if the init_ruri is not present for some reason we can not proceed with that
routine.
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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
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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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r284415 | tilghman | 2010-08-31 15:22:10 -0500 (Tue, 31 Aug 2010) | 21 lines
Merged revisions 284399 via svnmerge from
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r284399 | tilghman | 2010-08-31 15:18:32 -0500 (Tue, 31 Aug 2010) | 14 lines
Merged revisions 284393 via svnmerge from
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r284393 | tilghman | 2010-08-31 15:13:21 -0500 (Tue, 31 Aug 2010) | 7 lines
Don't send a devstate change on poke_noanswer if the state did not change.
(closes issue #17741)
Reported by: schmidts
Patches:
chan_sip.c.patch uploaded by schmidts (license 1077)
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r284318 | lmadsen | 2010-08-31 14:00:15 -0500 (Tue, 31 Aug 2010) | 22 lines
Merged revisions 284317 via svnmerge from
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r284317 | lmadsen | 2010-08-31 13:59:31 -0500 (Tue, 31 Aug 2010) | 15 lines
Merged revisions 284316 via svnmerge from
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r284316 | lmadsen | 2010-08-31 13:57:59 -0500 (Tue, 31 Aug 2010) | 7 lines
Update say.conf.sample to match the rules in say.c
(closes issue #17835)
Reported by: RoadKill
Patches:
say.conf.sample.patch.rules uploaded by RoadKill (license 933)
Tested by: RoadKill
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(closes issue #17860)
Reported by: jtodd
Patches:
__20100816-chan_sip-sip-show-settings.txt uploaded by lmadsen (license 10)
Tested by: lmadsen, russell
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r284281 | tilghman | 2010-08-30 17:28:47 -0500 (Mon, 30 Aug 2010) | 18 lines
Merged revisions 284280 via svnmerge from
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r284280 | tilghman | 2010-08-30 17:27:06 -0500 (Mon, 30 Aug 2010) | 11 lines
Fix 3 coding errors:
1) After we close FD, we should not be trying to write to it.
2) Call _exit(0), not exit(0), to avoid running shutdown routines in a child.
3) Use endian, not processor, detection to ensure bytes are written in the correct order.
(closes issue #15706)
Reported by: modelnine
Patches:
asterisk-1.6.1.1-festival-debug.patch uploaded by modelnine (license 865)
Tested by: gmartinez
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