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git.ipfire.org Git - thirdparty/asterisk.git/log
Jan Kalab [Tue, 14 Sep 2010 06:55:44 +0000 (06:55 +0000)]
Merging events for Exchange web service doesn't work as expected, resulting in only one event in calendar
The solution is to use "global" counter of events, since we do new requests for every event and calendar sync after every request. So now we do sync only after last request.
(closes issue #17877)
Review: https://reviewboard.asterisk.org/r/916/
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Tilghman Lesher [Tue, 14 Sep 2010 05:07:16 +0000 (05:07 +0000)]
Merged revisions 286587 via svnmerge from
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r286587 | tilghman | 2010-09-14 00:06:05 -0500 (Tue, 14 Sep 2010) | 2 lines
Add documentation on missing backend tables for Voicemail
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Tilghman Lesher [Mon, 13 Sep 2010 23:50:34 +0000 (23:50 +0000)]
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r286557 | tilghman | 2010-09-13 18:48:51 -0500 (Mon, 13 Sep 2010) | 2 lines
C precedence got me
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Tilghman Lesher [Mon, 13 Sep 2010 23:12:21 +0000 (23:12 +0000)]
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r286527 | tilghman | 2010-09-13 18:03:26 -0500 (Mon, 13 Sep 2010) | 2 lines
Refactor conversion to ast_poll() to fix callparking regression.
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Jason Parker [Mon, 13 Sep 2010 19:40:05 +0000 (19:40 +0000)]
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r286456 | qwell | 2010-09-13 14:38:35 -0500 (Mon, 13 Sep 2010) | 5 lines
Remove "Internal IP" from sip show settings, as it's not at all useful to display.
(closes issue #17840)
Reported by: oej
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Richard Mudgett [Mon, 13 Sep 2010 15:52:14 +0000 (15:52 +0000)]
Update chan_dahdi.conf.sample to reflect new libpri T309 default value.
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Olle Johansson [Sat, 11 Sep 2010 17:09:22 +0000 (17:09 +0000)]
Merged revisions 286268 via svnmerge from
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r286268 | oej | 2010-09-11 19:05:16 +0200 (Lör, 11 Sep 2010) | 11 lines
Merged revisions 286267 via svnmerge from
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r286267 | oej | 2010-09-11 18:59:20 +0200 (Lör, 11 Sep 2010) | 4 lines
Handle error response when we can't make file compatible
Review: https://reviewboard.asterisk.org/r/911/
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Terry Wilson [Fri, 10 Sep 2010 22:04:53 +0000 (22:04 +0000)]
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r286115 | twilson | 2010-09-10 15:35:25 -0500 (Fri, 10 Sep 2010) | 23 lines
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r286059 | twilson | 2010-09-10 14:25:08 -0500 (Fri, 10 Sep 2010) | 16 lines
Inherit CHANNEL() writes to both sides of a Local channel
Having Local (/n) channels as queue members and setting the language in the
extension with Set(CHANNEL(language)=fr) sets the language on the Local/...,2
channel. Hold time report playbacks happen on the Local/...,1 channel and
therefor do not play in the specified language.
This patch modifies func_channel_write to call the setoption callback and pass
the CHANNEL() write info to the callback. chan_local uses this information to
look up the other side of the channel and apply the same changes to it.
(closes issue #17673)
Reported by: Guggemand
Review: https://reviewboard.asterisk.org/r/903/
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Paul Belanger [Fri, 10 Sep 2010 21:11:08 +0000 (21:11 +0000)]
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r286117 | pabelanger | 2010-09-10 16:55:06 -0400 (Fri, 10 Sep 2010) | 11 lines
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r286114 | pabelanger | 2010-09-10 16:35:08 -0400 (Fri, 10 Sep 2010) | 4 lines
Load iax.conf before registering any functions/applications/actions.
Review: https://reviewboard.asterisk.org/r/914/
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Richard Mudgett [Fri, 10 Sep 2010 20:55:37 +0000 (20:55 +0000)]
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r286116 | rmudgett | 2010-09-10 15:42:44 -0500 (Fri, 10 Sep 2010) | 18 lines
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r286113 | rmudgett | 2010-09-10 15:33:16 -0500 (Fri, 10 Sep 2010) | 11 lines
An outgoing call may not get hung up if a pre-connect incoming ISDN call is disconnected.
If the ISDN link a pre-connect incoming call is using fails or is reset,
the outgoing leg may not hang up or be delayed in hanging up. (Causes:
PRI_CAUSE_NETWORK_OUT_OF_ORDER, PRI_CAUSE_DESTINATION_OUT_OF_ORDER, and
PRI_CAUSE_NORMAL_TEMPORARY_FAILURE.)
Just hang up the call if the incoming call leg hangs up before connecting
for any reason. It makes no sense to send a BUSY or CONGESTION control
frame to the outgoing call leg under these circumstances.
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Russell Bryant [Fri, 10 Sep 2010 20:31:58 +0000 (20:31 +0000)]
Rate limit calls to fsync() to 1 per second after astdb updates.
Astdb was determined to be one of the most significant bottlenecks in SIP
registration processing. This patch improved the speed of an astdb load
test by 50000% (yes, Fifty-Thousand Percent). On this particular load test
setup, this doubled the number of SIP registrations the server could handle.
Review: https://reviewboard.asterisk.org/r/825/
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Tilghman Lesher [Fri, 10 Sep 2010 18:31:12 +0000 (18:31 +0000)]
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r286024 | tilghman | 2010-09-10 13:30:21 -0500 (Fri, 10 Sep 2010) | 9 lines
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r286023 | tilghman | 2010-09-10 13:22:04 -0500 (Fri, 10 Sep 2010) | 2 lines
Missing newline
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David Ruggles [Fri, 10 Sep 2010 13:13:16 +0000 (13:13 +0000)]
Added missing documentation for ExternalIVR feature added in January 2010
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Tilghman Lesher [Fri, 10 Sep 2010 05:32:18 +0000 (05:32 +0000)]
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r285961 | tilghman | 2010-09-10 00:31:31 -0500 (Fri, 10 Sep 2010) | 6 lines
Another fix for Mac OS X.
While trying to fix this the "right" way, I wandered into dependency hell. Two
hours later, I backed out, and just removed the offending code. ast_inline_api
only goes one level deep and then it breaks. Ouch.
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Tilghman Lesher [Fri, 10 Sep 2010 01:25:50 +0000 (01:25 +0000)]
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r285930 | tilghman | 2010-09-09 20:16:32 -0500 (Thu, 09 Sep 2010) | 14 lines
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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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Paul Belanger [Thu, 9 Sep 2010 22:52:31 +0000 (22:52 +0000)]
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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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Jason Parker [Thu, 9 Sep 2010 20:11:06 +0000 (20:11 +0000)]
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r285744 | qwell | 2010-09-09 15:09:23 -0500 (Thu, 09 Sep 2010) | 16 lines
Merged revisions 285742 via svnmerge from
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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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Brett Bryant [Thu, 9 Sep 2010 18:51:52 +0000 (18:51 +0000)]
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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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Brett Bryant [Thu, 9 Sep 2010 17:23:28 +0000 (17:23 +0000)]
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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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David Vossel [Wed, 8 Sep 2010 22:14:19 +0000 (22:14 +0000)]
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r285567 | dvossel | 2010-09-08 17:11:28 -0500 (Wed, 08 Sep 2010) | 9 lines
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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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David Vossel [Wed, 8 Sep 2010 21:48:37 +0000 (21:48 +0000)]
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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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Brett Bryant [Wed, 8 Sep 2010 20:58:43 +0000 (20:58 +0000)]
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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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Jason Parker [Wed, 8 Sep 2010 20:43:10 +0000 (20:43 +0000)]
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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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Jason Parker [Wed, 8 Sep 2010 20:32:13 +0000 (20:32 +0000)]
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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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Tilghman Lesher [Wed, 8 Sep 2010 07:14:17 +0000 (07:14 +0000)]
Documentation only
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Jason Parker [Tue, 7 Sep 2010 22:22:14 +0000 (22:22 +0000)]
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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Tilghman Lesher [Tue, 7 Sep 2010 21:20:16 +0000 (21:20 +0000)]
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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Tilghman Lesher [Tue, 7 Sep 2010 21:14:03 +0000 (21:14 +0000)]
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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Richard Mudgett [Tue, 7 Sep 2010 21:08:35 +0000 (21:08 +0000)]
Fix cut-n-paste error.
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Jason Parker [Tue, 7 Sep 2010 20:58:34 +0000 (20:58 +0000)]
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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Tilghman Lesher [Tue, 7 Sep 2010 20:56:07 +0000 (20:56 +0000)]
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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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Tilghman Lesher [Tue, 7 Sep 2010 19:38:12 +0000 (19:38 +0000)]
Fix build on FreeBSD 8.0, take 2.
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Tilghman Lesher [Tue, 7 Sep 2010 19:08:09 +0000 (19:08 +0000)]
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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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Brett Bryant [Tue, 7 Sep 2010 17:54:21 +0000 (17:54 +0000)]
Merged revisions 285196 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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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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Richard Mudgett [Tue, 7 Sep 2010 17:47:34 +0000 (17:47 +0000)]
Merged revisions 285193 via svnmerge from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
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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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Russell Bryant [Mon, 6 Sep 2010 20:10:24 +0000 (20:10 +0000)]
regenerate configure script.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@285162
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Russell Bryant [Mon, 6 Sep 2010 20:10:03 +0000 (20:10 +0000)]
Fix libsrtp -fPIC check for when non-standard prefix is used.
Thanks to loompek in #asterisk for reporting the issue and testing this patch.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@285161
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Tilghman Lesher [Mon, 6 Sep 2010 06:56:07 +0000 (06:56 +0000)]
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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Russell Bryant [Sat, 4 Sep 2010 18:08:19 +0000 (18:08 +0000)]
Add a C++ compatible version of AST_CLI_DEFINE().
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@285057
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Terry Wilson [Fri, 3 Sep 2010 23:19:54 +0000 (23:19 +0000)]
Call correct lock function as transferer is a sip_pvt not a channel
Both functions are #defined to ao2_lock, but still...
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@285017
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David Vossel [Fri, 3 Sep 2010 22:21:50 +0000 (22:21 +0000)]
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@285006
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Brett Bryant [Fri, 3 Sep 2010 18:19:53 +0000 (18:19 +0000)]
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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David Vossel [Fri, 3 Sep 2010 18:03:23 +0000 (18:03 +0000)]
During OPTIONS authentication, the authpeer does not need to be returned for any reason.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284952
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David Vossel [Fri, 3 Sep 2010 17:29:02 +0000 (17:29 +0000)]
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284950
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Terry Wilson [Fri, 3 Sep 2010 16:28:18 +0000 (16:28 +0000)]
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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Jan Kalab [Fri, 3 Sep 2010 13:07:39 +0000 (13:07 +0000)]
Calendar categories and priorities: strdupa() fix
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284852
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Jan Kalab [Fri, 3 Sep 2010 12:48:59 +0000 (12:48 +0000)]
Fix for calendar categories and priorities according to ISO C90
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284850
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Jan Kalab [Fri, 3 Sep 2010 11:27:04 +0000 (11:27 +0000)]
Support for calendar events priorities and categories
Review 880
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284849
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Brett Bryant [Thu, 2 Sep 2010 21:04:51 +0000 (21:04 +0000)]
Merged revisions 284778 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r284778 | bbryant | 2010-09-02 16:54:33 -0400 (Thu, 02 Sep 2010) | 14 lines
Merged revisions 284777 via svnmerge from
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r284777 | bbryant | 2010-09-02 16:25:03 -0400 (Thu, 02 Sep 2010) | 7 lines
Fixes a bug in manager.c where the default configuration values weren't reset when the manager configuration was reloaded.
(closes issue #17917)
Reported by: lmadsen
Review: https://reviewboard.asterisk.org/r/883/
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Richard Mudgett [Thu, 2 Sep 2010 21:02:54 +0000 (21:02 +0000)]
Simplified pri_dchannel() poll timeout duration code.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284780
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Richard Mudgett [Thu, 2 Sep 2010 20:59:12 +0000 (20:59 +0000)]
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284779
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David Vossel [Thu, 2 Sep 2010 16:56:43 +0000 (16:56 +0000)]
Merged revisions 284704 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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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
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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Jason Parker [Thu, 2 Sep 2010 16:43:09 +0000 (16:43 +0000)]
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284701
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Richard Mudgett [Thu, 2 Sep 2010 16:34:32 +0000 (16:34 +0000)]
Added documentation for CONNECTEDLINE and REDIRECTING functions.
(closes issue #17808)
Reported by: jtodd
Review: https://reviewboard.asterisk.org/r/875/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284698
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Tilghman Lesher [Thu, 2 Sep 2010 16:27:52 +0000 (16:27 +0000)]
Fixing build
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284696
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Tilghman Lesher [Thu, 2 Sep 2010 16:11:15 +0000 (16:11 +0000)]
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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Tilghman Lesher [Thu, 2 Sep 2010 05:31:02 +0000 (05:31 +0000)]
Merged revisions 284631 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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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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Tilghman Lesher [Thu, 2 Sep 2010 05:20:59 +0000 (05:20 +0000)]
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284610
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Tilghman Lesher [Thu, 2 Sep 2010 05:00:34 +0000 (05:00 +0000)]
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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David Vossel [Wed, 1 Sep 2010 21:47:01 +0000 (21:47 +0000)]
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284561
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Terry Wilson [Wed, 1 Sep 2010 18:44:36 +0000 (18:44 +0000)]
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284477
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Tilghman Lesher [Wed, 1 Sep 2010 18:16:37 +0000 (18:16 +0000)]
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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Tilghman Lesher [Tue, 31 Aug 2010 20:22:10 +0000 (20:22 +0000)]
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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Leif Madsen [Tue, 31 Aug 2010 19:00:15 +0000 (19:00 +0000)]
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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Tilghman Lesher [Mon, 30 Aug 2010 22:28:47 +0000 (22:28 +0000)]
Merged revisions 284280 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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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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Tilghman Lesher [Sun, 29 Aug 2010 07:05:27 +0000 (07:05 +0000)]
Missed adding this file
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284158
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Tilghman Lesher [Sun, 29 Aug 2010 05:17:37 +0000 (05:17 +0000)]
Also ignore the checksums
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Tilghman Lesher [Sun, 29 Aug 2010 02:51:14 +0000 (02:51 +0000)]
Rename CEL adaptive driver to plain driver, since there isn't another ODBC driver
(and the other CEL drivers have adaptive capabilities, anyway).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284096
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Russell Bryant [Sat, 28 Aug 2010 21:29:45 +0000 (21:29 +0000)]
Be more flexible with whitespace on AMI action headers.
Previously, this code required exactly one space to be after the ':' in headers
for an AMI action. This now makes whitespace optional, and allows whitespace that
is there to vary in amount.
(closes issue #17862)
Reported by: cmoye
Patches:
manager.c.patch_trunk uploaded by cmoye (license 858)
manager.c.patch_1.8 uploaded by cmoye (license 858)
Tested by: cmoye
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284065
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David Vossel [Fri, 27 Aug 2010 22:37:11 +0000 (22:37 +0000)]
Merged revisions 284002 via svnmerge from
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r284002 | dvossel | 2010-08-27 17:27:50 -0500 (Fri, 27 Aug 2010) | 14 lines
Merged revisions 283960 via svnmerge from
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r283960 | dvossel | 2010-08-27 17:17:26 -0500 (Fri, 27 Aug 2010) | 8 lines
Parse all "Accept" headers for SIP SUBSCRIBE requests.
(closes issue #17758)
Reported by: ibc
Patches:
multiple_accept_headers_1.4.diff uploaded by dvossel (license 671)
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Russell Bryant [Fri, 27 Aug 2010 21:33:55 +0000 (21:33 +0000)]
Print exten@context:priority in verbose messages from pbx_realtime.
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Jason Parker [Fri, 27 Aug 2010 20:31:55 +0000 (20:31 +0000)]
Merged revisions 283881 via svnmerge from
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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
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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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Tilghman Lesher [Thu, 26 Aug 2010 23:47:02 +0000 (23:47 +0000)]
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@283770
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David Vossel [Thu, 26 Aug 2010 15:26:37 +0000 (15:26 +0000)]
Merged revisions 283691 via svnmerge from
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r283691 | dvossel | 2010-08-26 10:24:40 -0500 (Thu, 26 Aug 2010) | 25 lines
Merged revisions 283690 via svnmerge from
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r283690 | dvossel | 2010-08-26 10:22:28 -0500 (Thu, 26 Aug 2010) | 19 lines
Fixed how Asterisk destroys a dialog on channel hangup before invite receives a response.
If an ast_channel with a SIP tech pvt hangs up before the sip dialog gets a response
to its outgoing INVITE, Asterisk used to pretend_ack the INVITE. This is not rfc
compliant and results in confusion at the other endpoint. sip_pretend_ack will ack
and remove all the packets in the retransmit queue. This means that the INVITE will
stop retransmitting, and that any response to that INVITE that comes after the pretend_ack
occurs will be ignored.
Instead of faking any sort of acknowledgement for an outgoing INVITE during an internal
hangup, we should let the protocol stack process the INVITE transaction and terminate
the dialog properly. This is achieved by setting the PENDING_BYE flag. When this flag
is used, once the dialog proceeds to an escapable state the transaction will either be
canceled with a SIP_CANCEL or completed followed immediately by a BYE. Attempting to do
this any other way is incorrect. If the endpoint is not responding to the INVITE request,
the INVITE must continue to be retransmitted until it times out which will result in the
dialog being destroyed.
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Russell Bryant [Thu, 26 Aug 2010 13:26:14 +0000 (13:26 +0000)]
Slight improvement to a debug message.
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Russell Bryant [Thu, 26 Aug 2010 12:48:45 +0000 (12:48 +0000)]
Remove public keys that are no longer useful.
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Russell Bryant [Thu, 26 Aug 2010 12:26:22 +0000 (12:26 +0000)]
Move httptimeout out from in between port and bindaddr.
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David Vossel [Wed, 25 Aug 2010 22:57:56 +0000 (22:57 +0000)]
Merged revisions 283594 via svnmerge from
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r283594 | dvossel | 2010-08-25 17:56:42 -0500 (Wed, 25 Aug 2010) | 7 lines
Add to and from tags to NOTIFY dialog-info xml body so pickup can occur.
When pedantic mode is used, the dialog-info xml generated during a
ringing event must contain the to and from tag values. Otherwise if
a pickup occurs using INVITE with replaces, Astrisk will not be able
to locate the subscription.
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Tilghman Lesher [Wed, 25 Aug 2010 16:12:43 +0000 (16:12 +0000)]
Initialize connect timeout on each time through the loop.
(closes issue #17911)
Reported by: wurstsalat
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David Vossel [Wed, 25 Aug 2010 15:54:11 +0000 (15:54 +0000)]
Merged revisions 283558 via svnmerge from
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r283558 | dvossel | 2010-08-25 10:52:54 -0500 (Wed, 25 Aug 2010) | 10 lines
Asterisk will not advertise session timers are supported when 'session-timers=refuse' is used.
Asterisk now dynamically builds the "Supported" header depending
on what is enabled/disabled in sip.conf. Session timers used
to always be advertised as being supported even when they were disabled
in the configuration. This caused problems with some end points.
(issue #17005)
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Russell Bryant [Wed, 25 Aug 2010 14:55:00 +0000 (14:55 +0000)]
Convert ast_log(LOG_DEBUG, ...) to ast_debug(...)
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David Vossel [Tue, 24 Aug 2010 20:34:03 +0000 (20:34 +0000)]
Changes the default behavior for sip.conf's pedantic option from "no" to "yes".
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Leif Madsen [Tue, 24 Aug 2010 18:56:29 +0000 (18:56 +0000)]
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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David Vossel [Tue, 24 Aug 2010 16:11:18 +0000 (16:11 +0000)]
Merged revisions 283381 via svnmerge from
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r283381 | dvossel | 2010-08-24 11:07:37 -0500 (Tue, 24 Aug 2010) | 18 lines
Merged revisions 283380 via svnmerge from
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r283380 | dvossel | 2010-08-24 11:01:51 -0500 (Tue, 24 Aug 2010) | 11 lines
This fix makes sure the ast_channel hangs up correctly when the dialog's PENDING_BYE flag is set.
When the pending bye flag is used, it is possible that the dialog will terminate
and leave the sip_pvt->owner channel up. This is because we never hangup the
ast_channel after sending the SIP_BYE request. When we receive the response for
the SIP_BYE we set need_destroy which we would expect to destroy the dialog on the
next do_monitor loop, but this is not the case. The dialog will only be destroyed
once the owner is hungup even with the need_destroy flag set. This patch sets the
softhangup flag on the ast_channel when a SIP_BYE request is sent as a result of the
pending bye flag.
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Russell Bryant [Tue, 24 Aug 2010 12:49:41 +0000 (12:49 +0000)]
Don't attempt to release a NULL ODBC handle.
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Tilghman Lesher [Mon, 23 Aug 2010 21:33:47 +0000 (21:33 +0000)]
Merged revisions 283318 via svnmerge from
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r283318 | tilghman | 2010-08-23 16:32:14 -0500 (Mon, 23 Aug 2010) | 2 lines
CDR drivers depend upon res_odbc, not directly on the ODBC libraries
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Russell Bryant [Mon, 23 Aug 2010 13:35:35 +0000 (13:35 +0000)]
Add sample configuration for cel_radius.
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Russell Bryant [Mon, 23 Aug 2010 13:23:12 +0000 (13:23 +0000)]
Make the AST_CEL_AMA enum match up with the AST_CDR_ ama flag values.
Really, having 2 enums for this is silly and error prone, demonstrated by
the crash that I hit because there was an assumption in the code that the
values in each matched up. However, this is a quick fix to get them to
match up so it will work.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@283230
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Russell Bryant [Mon, 23 Aug 2010 13:06:57 +0000 (13:06 +0000)]
Don't blow up on an invalid AMA flag.
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Russell Bryant [Mon, 23 Aug 2010 12:31:20 +0000 (12:31 +0000)]
Tack on ${eventextra} to the sample cel_custom.conf.
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Russell Bryant [Mon, 23 Aug 2010 12:12:53 +0000 (12:12 +0000)]
Cut down on excessive quotation.
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Tilghman Lesher [Mon, 23 Aug 2010 12:06:26 +0000 (12:06 +0000)]
Don't fail to start if the config file is missing.
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Russell Bryant [Mon, 23 Aug 2010 11:58:34 +0000 (11:58 +0000)]
Expand cel_custom.conf.sample.
Include the usage of CSV_QUOTE() to ensure data has valid CSV formatting. Also list
the special CEL variables that are available for use in the mapping.
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Richard Mudgett [Fri, 20 Aug 2010 16:51:16 +0000 (16:51 +0000)]
Recorded merge of revisions 283124 via svnmerge from
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r283124 | rmudgett | 2010-08-20 11:48:10 -0500 (Fri, 20 Aug 2010) | 16 lines
Merged revisions 283123 via svnmerge from
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r283123 | rmudgett | 2010-08-20 11:46:22 -0500 (Fri, 20 Aug 2010) | 9 lines
Merged revision 278274 from
https://origsvn.digium.com/svn/asterisk/trunk
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r278274 | rmudgett | 2010-07-20 17:38:13 -0500 (Tue, 20 Jul 2010) | 1 line
Reference correct struct member for unlikely event PRI_EVENT_CONFIG_ERR.
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Richard Mudgett [Fri, 20 Aug 2010 15:35:38 +0000 (15:35 +0000)]
Merged revisions 283049 via svnmerge from
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r283049 | rmudgett | 2010-08-20 10:31:03 -0500 (Fri, 20 Aug 2010) | 29 lines
Merged revisions 283048 via svnmerge from
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r283048 | rmudgett | 2010-08-20 10:24:36 -0500 (Fri, 20 Aug 2010) | 22 lines
Q931 - Sending PROGRESS after sending ALERTING is a protocol error
The PRI layer in chan_dadhi will check if a PROGRESS message has already
been sent, and not allow sending another (although that is technically
allowed by the Q931 spec), however it does not protect against sending an
ALERTING and then sending a PROGRESS message, which is a violation of the
specification.
Most switches don't seem to care too deeply about this, but some do, and
will disconnect the call when receiving this invalid sequence.
Protocol specification reference: T-REC-Q.931-199805-I page 223, "Figure
A.5/Q.931 -- Overview protocol control (network side) point-point
(sheet 3 of 8)"
(closes issue #17874)
Reported by: nic_bellamy
Patches:
asterisk-1.4-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299)
asterisk-1.6.2-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299)
asterisk-trunk-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299)
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Russell Bryant [Fri, 20 Aug 2010 12:45:12 +0000 (12:45 +0000)]
Fix a typo in a column name.
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Russell Bryant [Fri, 20 Aug 2010 11:52:37 +0000 (11:52 +0000)]
Add an argument missing from the CELGenUserEvent documentation.
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David Vossel [Thu, 19 Aug 2010 21:07:20 +0000 (21:07 +0000)]
Merged revisions 282894 via svnmerge from
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r282894 | dvossel | 2010-08-19 16:05:54 -0500 (Thu, 19 Aug 2010) | 18 lines
Merged revisions 282893 via svnmerge from
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r282893 | dvossel | 2010-08-19 16:03:24 -0500 (Thu, 19 Aug 2010) | 11 lines
tos_sip option was not being set correctly
When tos_sip is used, the tos of the sip socket is only set
correctly if the socket binding changes on a reload. If the binding
stays the same but the TOS changes, the new tos value would not take
into effect. This patch fixes that.
(closes issue #17712)
Reported by: nickb
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David Vossel [Thu, 19 Aug 2010 20:34:41 +0000 (20:34 +0000)]
Merged revisions 282890 via svnmerge from
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r282890 | dvossel | 2010-08-19 15:31:22 -0500 (Thu, 19 Aug 2010) | 5 lines
fixes sip peer memory leaks in the peer_by_ip table
(issue #17798)
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Matthew Nicholson [Thu, 19 Aug 2010 20:01:11 +0000 (20:01 +0000)]
Merged revisions 282859 via svnmerge from
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r282859 | mnicholson | 2010-08-19 14:44:00 -0500 (Thu, 19 Aug 2010) | 23 lines
Merged revisions 277944 via svnmerge from
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r277944 | pabelanger | 2010-07-19 15:56:07 -0500 (Mon, 19 Jul 2010) | 16 lines
Regression with T.38 negotiation
Prior to 1.4.26.3 T.38 negotiation worked properly, in the case
of the reporter.
(issue #16852)
Reported by: cfc
(closes issue #16705)
Reported by: mpiazzatnetbug
Patches:
issue16705_2.diff uploaded by ebroad (license 878)
Tested by: vrban, ebroad, c0rnoTa, samdell3
Review: https://reviewboard.asterisk.org/r/754/
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projects / thirdparty / asterisk.git / log
