Tilghman Lesher [Wed, 7 Jul 2010 06:15:43 +0000 (06:15 +0000)]
Merged revisions 274417 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r274417 | tilghman | 2010-07-07 01:13:54 -0500 (Wed, 07 Jul 2010) | 8 lines
Correct how 100, 200, 300, etc. is said. Also add the crazy British numbers.
(closes issue #16102)
Reported by: Delvar
Patches:
say.conf.fix.patch uploaded by Delvar (license 908)
(plus a few additional fixes and simplifications by me)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274418
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Jeff Peeler [Tue, 6 Jul 2010 22:23:35 +0000 (22:23 +0000)]
Merged revisions 274283 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r274283 | jpeeler | 2010-07-06 17:15:21 -0500 (Tue, 06 Jul 2010) | 7 lines
Correct sip.conf.sample comments for prematuremedia option.
(closes issue #17513)
Reported by: festr
Patches:
patch uploaded by festr (license 443)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274316
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Terry Wilson [Tue, 6 Jul 2010 22:15:27 +0000 (22:15 +0000)]
Merged revisions 274280 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r274280 | twilson | 2010-07-06 17:08:20 -0500 (Tue, 06 Jul 2010) | 9 lines
Add option to not do a call forward on 482 Loop Detected
Asterisk has always set up a forwarded call when receiving a 482 Loop Detected.
This prevents handling the call failure by just continuing on in the dialplan.
Since this would be a change in behavior, the new option to disable this
behavior is forwardloopdetected which defaults to 'yes'.
Review: https://reviewboard.asterisk.org/r/764/
........
(no option for trunk, just changing the behavior)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274284
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Tilghman Lesher [Tue, 6 Jul 2010 22:09:23 +0000 (22:09 +0000)]
Status shows all non-CRC4 lines as "yellow", even if "yellow" was not in the bitfield.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274281
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Matthew Nicholson [Tue, 6 Jul 2010 19:53:04 +0000 (19:53 +0000)]
Properly detect and report invalid maxrate and maxrate values in the FAXOPT 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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274243
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Mark Michelson [Tue, 6 Jul 2010 14:31:13 +0000 (14:31 +0000)]
Merged revisions 274157 via svnmerge from
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274164
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Matthew Nicholson [Tue, 6 Jul 2010 13:52:56 +0000 (13:52 +0000)]
Blocked revisions 274093 via svnmerge
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r274093 | mnicholson | 2010-07-06 08:52:28 -0500 (Tue, 06 Jul 2010) | 2 lines
Make get_member_status return QUEUE_NO_MEMBERS instead of QUEUE_NO_REACHABLE_MEMBERS to make joinempty=no work again. This regression was introduced in 273639. Also fixed whitespace.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274094
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Tilghman Lesher [Tue, 6 Jul 2010 06:01:37 +0000 (06:01 +0000)]
Uh, yeah.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274053
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Tilghman Lesher [Mon, 5 Jul 2010 20:00:48 +0000 (20:00 +0000)]
Blocked revisions 273981 via svnmerge
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r273981 | tilghman | 2010-07-05 14:48:42 -0500 (Mon, 05 Jul 2010) | 2 lines
Command 'stop gracefully' doesn't.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273982
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Paul Belanger [Mon, 5 Jul 2010 13:53:44 +0000 (13:53 +0000)]
Merged revisions 273884 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r273884 | pabelanger | 2010-07-05 09:51:29 -0400 (Mon, 05 Jul 2010) | 8 lines
Remove extra line breaks from 'core show config mappings'
(closes issue #17583)
Reported by: pabelanger
Patches:
issue17583.patch uploaded by pabelanger (license 224)
Tested by: lmadsen
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273886
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Tilghman Lesher [Sat, 3 Jul 2010 02:36:31 +0000 (02:36 +0000)]
Merged revisions 273793 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r273793 | tilghman | 2010-07-02 16:36:39 -0500 (Fri, 02 Jul 2010) | 9 lines
Have the DEADLOCK_AVOIDANCE macro warn when an unlock fails, to help catch potentially large software bugs.
(closes issue #17407)
Reported by: pdf
Patches:
20100527__issue17407.diff.txt uploaded by tilghman (license 14)
Review: https://reviewboard.asterisk.org/r/751/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273830
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Tilghman Lesher [Fri, 2 Jul 2010 17:10:59 +0000 (17:10 +0000)]
Merged revisions 273717 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r273717 | tilghman | 2010-07-02 12:09:47 -0500 (Fri, 02 Jul 2010) | 8 lines
Autoservice loop optimization causes a busy loop, when channels are serviced while in hangup.
(closes issue #17564)
Reported by: ramonpeek
Patches:
20100630__issue17564.diff.txt uploaded by tilghman (license 14)
Tested by: ramonpeek
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273718
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Tilghman Lesher [Fri, 2 Jul 2010 16:57:50 +0000 (16:57 +0000)]
Blocked revisions 273639 via svnmerge
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r273639 | tilghman | 2010-07-02 10:46:27 -0500 (Fri, 02 Jul 2010) | 8 lines
If all members are paused, the wrong status is indicated.
(closes issue #17576)
Reported by: ramonpeek
Patches:
diff.txt uploaded by ramonpeek (license 266)
Tested by: ramonpeek
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273715
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Tilghman Lesher [Fri, 2 Jul 2010 16:57:28 +0000 (16:57 +0000)]
The switch fallthrough could create some errorneous situations, so best to force directly to the default case.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273714
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Tzafrir Cohen [Fri, 2 Jul 2010 15:57:02 +0000 (15:57 +0000)]
Fix various typos reported by Lintian
(Also fix the typos in the comments)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273641
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Russell Bryant [Thu, 1 Jul 2010 22:16:23 +0000 (22:16 +0000)]
Merged revisions 273565 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r273565 | russell | 2010-07-01 17:09:19 -0500 (Thu, 01 Jul 2010) | 7 lines
Don't return a partially initialized datastore.
If memory allocation fails in ast_strdup(), don't return a partially
initialized datastore. Bad things may happen.
(related to ABE-2415)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273566
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Jeff Peeler [Thu, 1 Jul 2010 20:28:15 +0000 (20:28 +0000)]
Merged revisions 273474 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r273474 | jpeeler | 2010-07-01 15:19:16 -0500 (Thu, 01 Jul 2010) | 14 lines
Allow admin user to join conference without using admin mode and no user pin.
Configuring the conference in meetme.conf like the following:
conf => 2345,,6666
did not prompt for pin when used without admin mode. This meant that the
conference could not be joined as an admin even if the user knew the correct
pin. The original bug report was submitted claiming that the blank user pin
should deny entry into the conference. I think a better way to handle this
would be with a feature enhancement that used the following syntax:
conf => 2345,X,6666 - where X denotes no acceptable pin allowed
(closes issue #15704)
Reported by: modelnine
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273522
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Matthew Nicholson [Thu, 1 Jul 2010 19:34:47 +0000 (19:34 +0000)]
Properly handle failures of fax->start_session()
FAX-177
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273464
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David Vossel [Thu, 1 Jul 2010 16:40:17 +0000 (16:40 +0000)]
correct handling of get_destination return values
A failure when calling the get_destination can mean multiple things. If
the extension is not found, a 404 error is appropriate, but if the URI
scheme is incorrect, a 404 is not approperiate. This patch adds the
get_destination_result enum to differentiate between these and other failure
types. The only logical difference in this patch is that we now send a "416
Unsupported URI scheme" response instead of a "404" when the scheme is not
recognized. This indicates to the initiator of the INVITE to retry the request
with a correct URI.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273427
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Jeff Peeler [Thu, 1 Jul 2010 15:12:31 +0000 (15:12 +0000)]
Merged revisions 273354 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r273354 | jpeeler | 2010-07-01 10:05:43 -0500 (Thu, 01 Jul 2010) | 12 lines
Ensure channel placed in meetme in ringing state is properly hung up.
An outgoing channel placed in meetme while still ringing which was then hung up
would not exit meetme and the channel was not properly destroyed. Specifically
checking for this scenario by looking at the appropriate control frames resolves
the issue.
(closes issue #15871)
Reported by: Ivan
Patches:
meetme_congestion_trunk_v2.patch uploaded by Ivan (license 229)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273355
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Matthew Nicholson [Thu, 1 Jul 2010 14:37:37 +0000 (14:37 +0000)]
Fixed whitespace problems
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273352
65c4cc65-6c06-0410-ace0-
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Matthew Nicholson [Thu, 1 Jul 2010 14:34:31 +0000 (14:34 +0000)]
Altered my comment about TCP_NODELAY
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273350
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Matthew Nicholson [Thu, 1 Jul 2010 12:57:18 +0000 (12:57 +0000)]
Don't free written frames in chan_mobile's mbl_write() function.
(closes issue #16430)
Reported by: azbest
Tested by: azbest
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273312
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Matthew Nicholson [Wed, 30 Jun 2010 18:48:21 +0000 (18:48 +0000)]
Set TCP_NODELAY on manager TCP sockets to prevent delays on outgoing packets. This regression was introduced in r48338.
AST-359
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273270
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Paul Belanger [Wed, 30 Jun 2010 17:28:04 +0000 (17:28 +0000)]
Fix rt(c)p set debug ip taking wrong argument
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273233
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Richard Mudgett [Wed, 30 Jun 2010 17:17:05 +0000 (17:17 +0000)]
Remove unnecessary if test in CV_DSTR()
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273198
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Richard Mudgett [Wed, 30 Jun 2010 17:15:46 +0000 (17:15 +0000)]
Misc doxygen cleanup in config.h
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273197
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Tilghman Lesher [Wed, 30 Jun 2010 01:07:02 +0000 (01:07 +0000)]
Permission checking for the system application is backwards.
(closes issue #17550)
Reported by: kenner
Patches:
manager.c.diff uploaded by kenner (license 1040)
Tested by: kenner
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273144
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Tilghman Lesher [Wed, 30 Jun 2010 01:01:14 +0000 (01:01 +0000)]
Don't attempt to proceed if our internal parser indicates an invalid file.
(closes issue #17560)
Reported by: Nick_Lewis
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273142
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Tilghman Lesher [Tue, 29 Jun 2010 23:20:40 +0000 (23:20 +0000)]
Merged revisions 273060 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r273060 | tilghman | 2010-06-29 18:15:28 -0500 (Tue, 29 Jun 2010) | 10 lines
Allow the "useragent" value to be restored into memory from the realtime backend.
This value is purely informational. It does not alter configuration at all.
(closes issue #16029)
Reported by: Guggemand
Patches:
realtime-useragent.patch uploaded by Guggemand (license 897)
Tested by: Guggemand
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273078
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Tilghman Lesher [Tue, 29 Jun 2010 22:59:51 +0000 (22:59 +0000)]
Recorded merge of revisions 273057 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r273057 | tilghman | 2010-06-29 17:58:58 -0500 (Tue, 29 Jun 2010) | 4 lines
_Really_ skip the channel... don't just retry for another 200 cycles.
(Closes issue SWP-1652, ABE-2240)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273058
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Tilghman Lesher [Tue, 29 Jun 2010 22:40:00 +0000 (22:40 +0000)]
Exclude libical for insufficient versions.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273055
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Tilghman Lesher [Tue, 29 Jun 2010 22:39:22 +0000 (22:39 +0000)]
Send DialPlanComplete as a response, not as a separate event.
Otherwise, it goes to all manager sessions and may exclude the current session,
if the Events mask excludes it.
(closes issue #17504)
Reported by: rrb3942
Patches:
showdialplan_patch.diff uploaded by rrb3942 (license 1003)
Tested by: rrb3942
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273054
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David Vossel [Tue, 29 Jun 2010 20:44:05 +0000 (20:44 +0000)]
send a 400 Bad Request on malformed sip request
RFC 2361 section 24.4.1 send a 400 Bad Request if the request
can not be understood due to malformed syntax. Currently we
simply ignore a packet with a missing callid, to, from, or
via header. Instead of ignoring we now send the 400 Bad request.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272981
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Tilghman Lesher [Mon, 28 Jun 2010 21:50:57 +0000 (21:50 +0000)]
Merged revisions 272925 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r272925 | tilghman | 2010-06-28 16:50:02 -0500 (Mon, 28 Jun 2010) | 8 lines
Don't change ownership/group/permissions on run directory, if it already exists.
(closes issue #17076)
Reported by: stuarth
Patches:
20100324__issue17076.diff.txt uploaded by tilghman (license 14)
Tested by: stuarth
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272926
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Tilghman Lesher [Mon, 28 Jun 2010 21:42:52 +0000 (21:42 +0000)]
Merged revisions 272921-272922 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r272921 | tilghman | 2010-06-28 16:29:27 -0500 (Mon, 28 Jun 2010) | 8 lines
Change the way that we read include files, to accommodate for changes in GCC 4.4.
(closes issue #17472)
Reported by: seandarcy
Patches:
config2.patch uploaded by nivan (license 1066)
Tested by: nivan
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r272922 | tilghman | 2010-06-28 16:38:49 -0500 (Mon, 28 Jun 2010) | 2 lines
Also trim trailing blanks on #includes
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David Vossel [Mon, 28 Jun 2010 18:38:47 +0000 (18:38 +0000)]
rfc compliant sip option parsing + new unit test
RFC 3261 section 8.2.2.3 states that if any unsupported options
are found in the Require header field, a "420 (Bad Extension)"
response should be sent with an Unsupported header field containing
only the unsupported options.
This is not currently being done correctly. Right now, if Asterisk
detects any unsupported sip options in a Require header the entire
list of options are returned in the Unsupported header even if some
of those options are in fact supported. This patch fixes that by
building an unsupported options character buffer when parsing the
options that can be sent with the 420 response. A unit test verifying
this functionality has been created. Some code refactoring was required.
Review: https://reviewboard.asterisk.org/r/680/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272880
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Mark Michelson [Mon, 28 Jun 2010 17:33:12 +0000 (17:33 +0000)]
Merged revisions 272804 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r272804 | mmichelson | 2010-06-28 12:31:40 -0500 (Mon, 28 Jun 2010) | 5 lines
Decode URI in contact header of 302 response.
ABE-2352
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272805
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Russell Bryant [Mon, 28 Jun 2010 15:33:32 +0000 (15:33 +0000)]
Use the underscore package so that underscores do not need to be escaped.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272684
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David Vossel [Mon, 28 Jun 2010 14:55:25 +0000 (14:55 +0000)]
code guidelines cleanup for retrans_pkt() function
I am doing work in this function. I noticed a large number of
coding guidline fixes that needed to be made. Rather than have
those changes distract from my functional changes I decided
to separate these into a separate patch.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272652
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Tilghman Lesher [Fri, 25 Jun 2010 20:18:47 +0000 (20:18 +0000)]
Merged revisions 272562 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r272562 | tilghman | 2010-06-25 15:17:37 -0500 (Fri, 25 Jun 2010) | 5 lines
Make the structure of the table specified before match the queries and results.
(closes issue #17557)
Reported by: cmaj
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272568
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Matthew Nicholson [Fri, 25 Jun 2010 19:42:54 +0000 (19:42 +0000)]
Implemement support for handling multiple documents when sending.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272558
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David Vossel [Fri, 25 Jun 2010 19:39:53 +0000 (19:39 +0000)]
chan_sip: more accurate retransmissions
RFC3261 states that Timer A should start at 500ms (T1) by default.
In chan_sip this value initially started at 1000ms and I changed
it to 500ms recently. After doing that I noticed in my packet
captures that it still occasionally retransmitted starting at
1000ms instead of 500ms like I told it to. This occurs because
the scheduler runs in the do_monitor thread. If a new retransmission
is added while the do_monitor thread is sleeping then it may not
detect that retransmission for nearly 1000ms. To fix this I just
poke the do_monitor thread to wake up when a new packet is sent
reliably requiring retransmits. The thread then detects the new
scheduler entry and adjusts its sleep time to account for it.
Review: https://reviewboard.asterisk.org/r/747
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272557
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Tilghman Lesher [Fri, 25 Jun 2010 19:17:16 +0000 (19:17 +0000)]
Symlink sounds files, to save disk space, when multiple tarballs/checkouts are on the same system.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272533
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Richard Mudgett [Thu, 24 Jun 2010 22:11:26 +0000 (22:11 +0000)]
Merged revisions 272446 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r272446 | rmudgett | 2010-06-24 16:58:49 -0500 (Thu, 24 Jun 2010) | 10 lines
ss_thread calls pri_grab without lock during overlap dial
Recent changes to chan_dahdi with relation to overlap dialing call
pri_grab without first obtaining a lock.
(closes issue #17414)
Reported by: pdf
Patches:
bug17414.patch uploaded by jpeeler (license 325)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272447
65c4cc65-6c06-0410-ace0-
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Russell Bryant [Wed, 23 Jun 2010 23:09:28 +0000 (23:09 +0000)]
Resolve some errors produced during module unload of chan_iax2.
The external test suite stops Asterisk using the "core stop gracefully" command.
The logs from the tests show that there are a number of problems with Asterisk
trying to cleanly shut down. This patch addresses the following type of error
that comes from chan_iax2:
[Jun 22 16:58:11] ERROR[29884]: lock.c:129 __ast_pthread_mutex_destroy:
chan_iax2.c line 11371 (iax2_process_thread_cleanup):
Error destroying mutex &thread->lock: Device or resource busy
For an example in the context of a build, see:
http://bamboo.asterisk.org/browse/AST-TRUNK-739/log
The primary purpose of this patch is to change the thread pool shutdown
procedure to be more explicit to ensure that the thread exits from a point
where it is not holding a lock. While testing that, I encountered various
crashes due to the order of operations in unload_module() being problematic.
I reordered some things there, as well.
Review: https://reviewboard.asterisk.org/r/736/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272370
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Matthew Nicholson [Wed, 23 Jun 2010 22:36:49 +0000 (22:36 +0000)]
Merged revisions 272367 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
This version of the patch only adds AgentComplete for attended transfers. It was already present for blind transfers.
........
r272367 | mnicholson | 2010-06-23 17:33:51 -0500 (Wed, 23 Jun 2010) | 8 lines
Send AgentComplete manager events in the event of blind and attended transfers.
(closes issue #16819)
Reported by: elbriga
Patches:
app_queue.diff uploaded by elbriga (license 482)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272368
65c4cc65-6c06-0410-ace0-
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Tilghman Lesher [Wed, 23 Jun 2010 21:53:49 +0000 (21:53 +0000)]
If there is realtime configuration, it does not get re-read on reload unless 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: https://origsvn.digium.com/svn/asterisk/trunk@272332
65c4cc65-6c06-0410-ace0-
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Tilghman Lesher [Wed, 23 Jun 2010 21:06:40 +0000 (21:06 +0000)]
Ensure a NULL file while debugging cannot crash AEL.
(closes issue #17215)
Reported by: vazir
Patches:
20100518__issue17215.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272260
65c4cc65-6c06-0410-ace0-
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Paul Belanger [Wed, 23 Jun 2010 21:06:15 +0000 (21:06 +0000)]
Fix previous merge. ast_test_flag != ast_test_flag64
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272259
65c4cc65-6c06-0410-ace0-
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Paul Belanger [Wed, 23 Jun 2010 21:00:00 +0000 (21:00 +0000)]
Merged revisions 272255 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r272255 | pabelanger | 2010-06-23 16:57:01 -0400 (Wed, 23 Jun 2010) | 12 lines
First caller into a dynamic conference now enter pin once.
If MeetMe is configured to use dynamic conference
numbers, then the first caller (which creates the
conference) had to enter the PIN number twice.
(closes issue #15878)
Reported by: shawkris
Patches:
issue15878.patch uploaded by pabelanger (license 224)
Tested by: pabelanger
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272257
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Terry Wilson [Wed, 23 Jun 2010 20:59:17 +0000 (20:59 +0000)]
Update configure when changing autconf m4 files...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272256
65c4cc65-6c06-0410-ace0-
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Terry Wilson [Wed, 23 Jun 2010 20:53:48 +0000 (20:53 +0000)]
Honor the --with-${library}=path for AST_EXT_TOOL_CHECK
(closes issue #16991)
Reported by: pprindeville
Patches:
with_netsnmp.patch.txt uploaded by twilson (license 396)
Tested by: twilson
Review: https://reviewboard.asterisk.org/r/739/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272254
65c4cc65-6c06-0410-ace0-
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Paul Belanger [Wed, 23 Jun 2010 20:35:45 +0000 (20:35 +0000)]
Correct manager variable 'EventList' case.
(closes issue #17520)
Reported by: kobaz
Patches:
manager.patch uploaded by kobaz (license 834)
Tested by: lmadsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272252
65c4cc65-6c06-0410-ace0-
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Paul Belanger [Wed, 23 Jun 2010 20:22:44 +0000 (20:22 +0000)]
Add localization support for Spanish
(closes issue #17548)
Reported by: cjacobsen
Patches:
say.conf.sample.diff uploaded by cjacobsen (license 1029)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272243
65c4cc65-6c06-0410-ace0-
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Tim Ringenbach [Wed, 23 Jun 2010 19:59:43 +0000 (19:59 +0000)]
Add new AMI command LocalOptimizeAway.
This command lets you request a "/n" local channel
optimize itself out of the way anyway.
Review: https://reviewboard.asterisk.org/r/732/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272218
65c4cc65-6c06-0410-ace0-
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Tilghman Lesher [Wed, 23 Jun 2010 18:45:18 +0000 (18:45 +0000)]
D'oh! Defaultenabled FTL.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272150
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Tilghman Lesher [Wed, 23 Jun 2010 18:41:18 +0000 (18:41 +0000)]
Recorded merge of revisions 272147 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r272147 | tilghman | 2010-06-23 13:40:28 -0500 (Wed, 23 Jun 2010) | 5 lines
Backport part of revision 136715 to fix callerid in voicemail text files (IMAP only).
(closes issue #16945)
Reported by: mneuhauser
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272148
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Terry Wilson [Wed, 23 Jun 2010 18:39:20 +0000 (18:39 +0000)]
Don't start the sla thread unless we realy need it
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272146
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Tilghman Lesher [Wed, 23 Jun 2010 18:25:54 +0000 (18:25 +0000)]
Load all lines from realtime, not just the first one.
(closes issue #17144)
Reported by: nahuelgreco
Patches:
20100513__issue17144__trunk.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272145
65c4cc65-6c06-0410-ace0-
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Terry Wilson [Wed, 23 Jun 2010 17:21:40 +0000 (17:21 +0000)]
Make sure reload updates SLA config
Even if there are no stations or trunks defined, we need to start the sla
thread to make sure we get the reload event. Also, when doing a reload we need
to remove the existing trunks and stations or they end up hanging around.
(closes issue #16818)
Reported by: mbonin
Patches:
sla_reload.patch uploaded by twilson (license 396)
Tested by: twilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272109
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Mark Michelson [Wed, 23 Jun 2010 17:08:34 +0000 (17:08 +0000)]
Add extra protection for reinvite glare scenario.
Testing proved that if Asterisk sent a connected line reinvite, and
the endpoint to which the reinvite were being sent sent a reinvite, Asterisk
would not properly respond with a 491 response.
The reason is that on connected line reinvites, we set the dialog's invitestate
to INV_CALLING to prevent Asterisk from sending a rapid flurry of connected line
reinvites. For other reinvites we do not do this. Because of the current invitestate,
when Asterisk received the reinvite, we interpreted this as a spiraled INVITE, and thus
did not behave properly.
The fix for this is to not enter the loop detection or spiral logic in handle_request_invite
if the channel state is currently up. This way, no mid-call reinvites will be misinterpreted,
no matter what the nature of the reinvite may have been.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272090
65c4cc65-6c06-0410-ace0-
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Russell Bryant [Tue, 22 Jun 2010 23:20:37 +0000 (23:20 +0000)]
Don't try to lock/unlock an uninitialized lock on a dahdi_pri.
This small changes prevents destroy_all_channels() from accessing a lock on an
unused dahdi_pri struct, resolving a ton of ERRORs that get spewed out when
shutting Asterisk down gracefully.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272052
65c4cc65-6c06-0410-ace0-
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David Vossel [Tue, 22 Jun 2010 22:11:50 +0000 (22:11 +0000)]
fixes issue with 'dialplan remove extension blah' segfaulting with tab completion
(closes issue #17440)
Reported by: kobaz
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272014
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David Vossel [Tue, 22 Jun 2010 20:37:05 +0000 (20:37 +0000)]
ignore CANCEL request after having already received final response to INVITE
RFC 3261 section 9 states that a CANCEL has no effect on a
request to a UAS that has already given a final response. This
patch checks to make sure there is a pending invite before
allowing a CANCEL request to be processed, otherwise it responds
to the CANCEL with a "481 Call/Transaction Does Not Exist".
Review: https://reviewboard.asterisk.org/r/697/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271977
65c4cc65-6c06-0410-ace0-
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David Vossel [Tue, 22 Jun 2010 17:57:28 +0000 (17:57 +0000)]
minor fixes for white/black event filters
This fixes a ref count leak in event filters and checks for
a filter container allocation failure during session creation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271905
65c4cc65-6c06-0410-ace0-
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Matthew Nicholson [Tue, 22 Jun 2010 17:35:17 +0000 (17:35 +0000)]
Merged revisions 271902 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r271902 | mnicholson | 2010-06-22 12:31:57 -0500 (Tue, 22 Jun 2010) | 8 lines
Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is set. This is necessary to keep the ref count correct.
(closes issue #16815)
Reported by: rain
Patches:
chan_sip-unref-fix.diff uploaded by rain (license 327) (modified)
Tested by: rain
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271903
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Jeff Peeler [Tue, 22 Jun 2010 16:29:18 +0000 (16:29 +0000)]
Add regular expression filtering for manager events.
This patch as documented in the sample config allows one to optionally apply
white, black, or both types of filtering to manager events. The new
'eventfilter' option is set per user.
(closes issue #14861)
Reported by: fnordian
Patches:
eventfilter3.patch uploaded by fnordian (license 110),
modified by me
Review: https://reviewboard.asterisk.org/r/673/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271868
65c4cc65-6c06-0410-ace0-
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Russell Bryant [Tue, 22 Jun 2010 16:28:03 +0000 (16:28 +0000)]
Resolve some errors that occur on a graceful shutdown.
Don't Finalize() if Initialize() did not succeed. This resulted in an error
about trying to Finalize() an invalid handle.
Also trim some trailing whitespace while in the area.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271867
65c4cc65-6c06-0410-ace0-
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Russell Bryant [Tue, 22 Jun 2010 16:17:14 +0000 (16:17 +0000)]
Change the method of retrieving the Asterisk version string.
Using this method makes it so res_fax doesn't have to be rebuilt on every
svn update.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271833
65c4cc65-6c06-0410-ace0-
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David Vossel [Tue, 22 Jun 2010 15:46:22 +0000 (15:46 +0000)]
fixes attended transfer behavior when both transferee and transferer hung up
If both the transferer and transferee of a attended transfer hangup before
the new channel picks up, the new channel should be hung up as well as it
has no endpoint to talk to. This mirrors the expected behavior used in 1.4.
(closes issue #17444)
Reported by: corruptor
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271831
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Matthew Nicholson [Tue, 22 Jun 2010 15:08:39 +0000 (15:08 +0000)]
Updated the CHANGES file documenting the addition of a configurable port in the dundi config file.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271764
65c4cc65-6c06-0410-ace0-
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Matthew Nicholson [Tue, 22 Jun 2010 14:54:58 +0000 (14:54 +0000)]
Merged revisions 271761 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r271761 | mnicholson | 2010-06-22 09:49:36 -0500 (Tue, 22 Jun 2010) | 9 lines
Allow users to specify a port for dundi peers.
(closes issue #17056)
Reported by: klaus3000
Patches:
dundi-peerport-patch-trunk.txt uploaded by klaus3000 (license 65)
Tested by: klaus3000
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271762
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Matthew Nicholson [Tue, 22 Jun 2010 12:58:28 +0000 (12:58 +0000)]
Merged revisions 271689 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r271689 | mnicholson | 2010-06-22 07:52:27 -0500 (Tue, 22 Jun 2010) | 8 lines
Modify chan_sip's packet generation api to automatically calculate the Content-Length. This is done by storing packet content in a buffer until it is actually time to send the packet, at which time the size of the packet is calculated. This change was made to ensure that the Content-Length is always correct.
(closes issue #17326)
Reported by: kenner
Tested by: mnicholson, kenner
Review: https://reviewboard.asterisk.org/r/693/
........
This change also adds an ast_str_copy_string() function (similar to ast_copy_string), that copies one ast_str into another, properly handling embedded nulls.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271690
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Tilghman Lesher [Mon, 21 Jun 2010 22:41:00 +0000 (22:41 +0000)]
Conflict kqueue on OS X, since it doesn't work there yet, anyway.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271657
65c4cc65-6c06-0410-ace0-
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David Vossel [Mon, 21 Jun 2010 21:58:33 +0000 (21:58 +0000)]
add speex 16khz sample frame so codec cost can be calculated
(closes issue #17534)
Reported by: fabled
Patches:
speex-wb-sample.diff uploaded by fabled (license 448)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271625
65c4cc65-6c06-0410-ace0-
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Jeff Peeler [Mon, 21 Jun 2010 20:46:53 +0000 (20:46 +0000)]
Merged revisions 271552 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r271552 | jpeeler | 2010-06-21 15:37:47 -0500 (Mon, 21 Jun 2010) | 7 lines
Do not use sizeof to calculate size of a heap allocated character array.
Change left out from 271399.
(closes issue #16053)
Reported by: diLLec
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271554
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David Vossel [Mon, 21 Jun 2010 20:46:22 +0000 (20:46 +0000)]
fixes crash when From header URI is missing "sip:"
(closes issue #17437)
Reported by: klaus3000
Patches:
sip_crash uploaded by dvossel (license 671)
Tested by: klaus3000
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271553
65c4cc65-6c06-0410-ace0-
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David Vossel [Mon, 21 Jun 2010 20:33:41 +0000 (20:33 +0000)]
fixes logic error introduced by slin16 sip support
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271551
65c4cc65-6c06-0410-ace0-
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Tilghman Lesher [Mon, 21 Jun 2010 05:10:06 +0000 (05:10 +0000)]
Add new application for declining counting words in multiple languages.
(closes issue #16869)
Reported by: chappell
Patches:
app_say_counted-
20100317.c uploaded by chappell (license 8)
Tested by: chappell
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271520
65c4cc65-6c06-0410-ace0-
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Jeff Peeler [Fri, 18 Jun 2010 21:32:09 +0000 (21:32 +0000)]
Merged revisions 271399 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r271399 | jpeeler | 2010-06-18 14:28:24 -0500 (Fri, 18 Jun 2010) | 11 lines
Fix crash when parsing some heavily nested statements in AEL on reload.
Due to the recursion used when compiling AEL in gen_prios, all the stack space
was being consumed when parsing some AEL that contained nesting 13 levels deep.
Changing a few large buffers to be heap allocated fixed the crash, although I
did not test how many more levels can now be safely used.
(closes issue #16053)
Reported by: diLLec
Tested by: jpeeler
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271483
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David Vossel [Fri, 18 Jun 2010 18:59:05 +0000 (18:59 +0000)]
file.c was truncating audio file formats to the lower 32bits.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271341
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Jeff Peeler [Fri, 18 Jun 2010 18:36:55 +0000 (18:36 +0000)]
Recorded merge of revisions 271335 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r271335 | jpeeler | 2010-06-18 13:33:17 -0500 (Fri, 18 Jun 2010) | 13 lines
Eliminate deadlock potential in dahdi_fixup().
(This is a backport of 269307, committed to trunk by rmudgett.)
Calling dahdi_indicate() when the channel private lock is already
held can cause a deadlock if the PRI lock is needed because
dahdi_indicate() will also get the channel private lock. The pri_grab()
function assumes that the channel private lock is held once to avoid
deadlock.
(closes issue #17261)
Reported by: aragon
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271336
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David Vossel [Thu, 17 Jun 2010 21:23:41 +0000 (21:23 +0000)]
fixes some coding guideline issue
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271300
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David Vossel [Thu, 17 Jun 2010 18:45:32 +0000 (18:45 +0000)]
retransmit response to BYE requests until timer J expires
According to RFC 3261 section 17.2.2, which describes non-INVITE server
transaction, when a dialog enters the Completed state it must destroy
the dialog after Timer J (T1*64) fires. For a BYE transaction Asterisk
terminates the dialog immediately during sip_hangup() when it should be
waiting T1*64 ms. This results in some odd behavior. For instance if
Asterisk receives a BYE and transmits a 200ok in response, if the endpoint
never receives the 200ok it will retransmit the BYE to which Asterisk
responds with a "481 Call leg/transaction does not exist" because the
dialog is already gone.
To resolve this I made a function called sip_scheddestroy_final(). This
differs slightly from sip_schedestroy() in that it enables a flag that
will prevent the destruction from ever being rescheduled or canceled
afterwards. It also prevents the pvt's needdestroy flag from being set
which triggers the destruction of the dialog within the do_monitor thread().
By using this function we are guaranteed destruction will not occur
until the scheduled time. This allows Asterisk to respond to any possible
retransmits for a dialog after we process the initial BYE request for T1*64 ms.
Other changes: I removed two instances where sip_cancel_destroy is used
right before calling sip_scheddestroy. sip_scheddestroy always calls
sip_cancel_destroy before scheduling the new destruction so it is completely
unnecessary.
Review: https://reviewboard.asterisk.org/r/694/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271262
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David Vossel [Thu, 17 Jun 2010 18:36:06 +0000 (18:36 +0000)]
adds support for slin16 in sip
(closes issue #16153)
Reported by: kfister
Patches:
16153-1.6.2.0-rc5.patch uploaded by kfister (license 912)
slin16.sip.patch.1 uploaded by malcolmd (license 924)
Tested by: kfister, malcolmd
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271261
65c4cc65-6c06-0410-ace0-
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David Vossel [Thu, 17 Jun 2010 17:23:43 +0000 (17:23 +0000)]
adds speex 16khz audio support
(closes issue #17501)
Reported by: fabled
Patches:
asterisk-trunk-speex-wideband-v2.patch uploaded by fabled (license 448)
Tested by: malcolmd, fabled, dvossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271231
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Jeff Peeler [Thu, 17 Jun 2010 15:34:08 +0000 (15:34 +0000)]
Change expected operation from error to debug message
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271192
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Matthew Nicholson [Thu, 17 Jun 2010 15:11:55 +0000 (15:11 +0000)]
Blocked revisions 271123 via svnmerge
........
r271123 | mnicholson | 2010-06-17 10:11:27 -0500 (Thu, 17 Jun 2010) | 7 lines
Set sin_family in ast_get_ip_or_srv() and removed the 'last' member of the ast_dnsmgr_entry struct.
(closes issue #15827)
Reported by: DennisD
Patches:
(modified) dnsmgr_15827.patch uploaded by chappell (license 8)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271124
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Paul Belanger [Thu, 17 Jun 2010 00:30:51 +0000 (00:30 +0000)]
option w[(secs)] incorrectly capitalized in xmldoc
(closes issue #17516)
Reported by: karlfife
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David Vossel [Wed, 16 Jun 2010 22:37:45 +0000 (22:37 +0000)]
addition of more parse_uri test cases
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271056
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Paul Belanger [Wed, 16 Jun 2010 21:17:39 +0000 (21:17 +0000)]
Merged revisions 270979 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r270979 | pabelanger | 2010-06-16 17:10:05 -0400 (Wed, 16 Jun 2010) | 4 lines
Fixed typo in macro-page
Reported to #asterisk-dev by a student of jsmith.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270987
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Jason Parker [Wed, 16 Jun 2010 21:12:25 +0000 (21:12 +0000)]
Fix the actual place that was pointed out, for previous commit.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270983
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Jason Parker [Wed, 16 Jun 2010 21:10:48 +0000 (21:10 +0000)]
Merged revisions 270980 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r270980 | qwell | 2010-06-16 16:10:09 -0500 (Wed, 16 Jun 2010) | 4 lines
Need to lock the agent chan before access its internal bits.
Pointed out by russellb on asterisk-dev mailing list.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270981
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Matthew Nicholson [Wed, 16 Jun 2010 20:34:31 +0000 (20:34 +0000)]
Set sin_family to AF_INET when doing lookups, also reset sin_port the first time the ip address changes.
(closes issue #17496)
Reported by: ManChicken
(closes issue #15827)
Reported by: DennisD
Patches:
dnsmgr_15827.patch uploaded by chappell (license 8)
Tested by: DennisD, gentlec, damage, wimpy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270974
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David Vossel [Wed, 16 Jun 2010 19:03:24 +0000 (19:03 +0000)]
addition of G.719 pass-through support
(closes issue #16293)
Reported by: malcolmd
Patches:
g719.passthrough.patch.7 uploaded by malcolmd (license 924)
format_g719.c uploaded by malcolmd (license 924)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270940
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Paul Belanger [Wed, 16 Jun 2010 18:43:22 +0000 (18:43 +0000)]
MSG_OOB flag on HANGUP packet removed.
Per Tilghman's request on IRC (#asterisk-bugs).
(closes issue #17506)
Reported by: brycebaril
Tested by: pabelanger, tilghman
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270936
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David Vossel [Wed, 16 Jun 2010 17:36:51 +0000 (17:36 +0000)]
Merged revisions 270866 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r270866 | dvossel | 2010-06-16 12:35:29 -0500 (Wed, 16 Jun 2010) | 22 lines
fixes chan_iax2 race condition
There is code in chan_iax2.c that attempts to guarantee that only a single
active thread will handle a call number at a time. This code works once
the thread is added to an active_list of threads, but we are not currently
guaranteed that a newly activated thread will enter the active_list immediately
because it is left up to the thread to add itself after frames have been
queued to it. This means that if two frames come in for the same call number
at the same time, it is possible for them to grab two separate threads because
the first thread did not add itself to the active_list fast enough. This
causes some pretty complex problems.
This patch resolves this race condition by immediately adding an activated
thread to the active_list within the network thread and only depending on
the thread to remove itself once it is done processing the frames queued to
it. By doing this we are guaranteed that if another frame for the same call
number comes in at the same time, that this thread will immediately be found
in the active_list of threads.
Review: https://reviewboard.asterisk.org/r/720/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270867
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Jeff Peeler [Wed, 16 Jun 2010 16:45:07 +0000 (16:45 +0000)]
Fix no call waiting caller ID
Clearing the callwaitcas flag in analog_call was causing the incoming D digit
to be ignored which triggers sending the caller ID.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270836
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Paul Belanger [Wed, 16 Jun 2010 15:05:11 +0000 (15:05 +0000)]
Update formatting for channelvariables.tex
(closes issue #17511)
Reported by: klaus3000
Patches:
channelvariables.tex-patch.txt uploaded by klaus3000 (license 65)
Tested by: pabelanger
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270801
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