Tilghman Lesher [Sat, 2 Oct 2010 04:54:13 +0000 (04:54 +0000)]
Merged revisions 289875 via svnmerge from
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r289875 | tilghman | 2010-10-01 23:46:43 -0500 (Fri, 01 Oct 2010) | 22 lines
Merged revisions 289874 via svnmerge from
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r289874 | tilghman | 2010-10-01 23:45:49 -0500 (Fri, 01 Oct 2010) | 15 lines
Merged revisions 289873 via svnmerge from
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r289873 | tilghman | 2010-10-01 23:42:08 -0500 (Fri, 01 Oct 2010) | 8 lines
When forwarding a message, a prepend means that the filesystem will always have a better copy.
(closes issue #17803)
Reported by: dpetersen
Patches:
20100923__issue17803.diff.txt uploaded by tilghman (license 14)
Tested by: dpetersen
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Jeff Peeler [Sat, 2 Oct 2010 02:46:43 +0000 (02:46 +0000)]
Merged revisions 289840 via svnmerge from
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r289840 | jpeeler | 2010-10-01 21:43:45 -0500 (Fri, 01 Oct 2010) | 29 lines
Merged revisions 289798 via svnmerge from
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r289798 | jpeeler | 2010-10-01 18:01:31 -0500 (Fri, 01 Oct 2010) | 22 lines
Merged revisions 289797 via svnmerge from
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r289797 | jpeeler | 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010) | 15 lines
Change RFC2833 DTMF event duration on end to report actual elapsed time.
The scenario here is with a non P2P early media session. The reported time
length of DTMF presses are coming up short when sending to the remote side.
Currently the event duration is a running total that is incremented when sending
continuation packets. These continuation packets are only triggered upon
incoming media from the remote side, which means that the running total probably
is not going to end up matching the actual length of time Asterisk received
DTMF. This patch changes the end event duration to be lengthened if it is
detected that the end event is going to come up short.
Review: https://reviewboard.asterisk.org/r/957/
ABE-2476
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Paul Belanger [Fri, 1 Oct 2010 17:22:30 +0000 (17:22 +0000)]
Merged revisions 289718 via svnmerge from
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r289718 | pabelanger | 2010-10-01 13:19:49 -0400 (Fri, 01 Oct 2010) | 20 lines
Merged revisions 289704 via svnmerge from
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r289704 | pabelanger | 2010-10-01 13:09:03 -0400 (Fri, 01 Oct 2010) | 13 lines
Merged revisions 289703 via svnmerge from
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r289703 | pabelanger | 2010-10-01 13:03:11 -0400 (Fri, 01 Oct 2010) | 6 lines
Disable debugging by default
and reformat .config file.
Review: https://reviewboard.asterisk.org/r/929/
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Jeff Peeler [Fri, 1 Oct 2010 16:23:16 +0000 (16:23 +0000)]
Merged revisions 289701 via svnmerge from
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r289701 | jpeeler | 2010-10-01 11:22:19 -0500 (Fri, 01 Oct 2010) | 28 lines
Merged revisions 289700 via svnmerge from
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r289700 | jpeeler | 2010-10-01 11:21:04 -0500 (Fri, 01 Oct 2010) | 21 lines
Merged revisions 289699 via svnmerge from
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r289699 | jpeeler | 2010-10-01 11:20:00 -0500 (Fri, 01 Oct 2010) | 14 lines
Ensure user portion of SIP URI matches dialplan when using encoded characters.
This commit takes a simliar approach to 288112 and checks the dialplan to
determine the proper action for an incoming contact header as to whether or not
it should be decoded or not. sip_new was blindly always decoding the extension,
which also caused the outgoing contact header to be incorrect as well as failing
to match the encoded extension in the dialplan.
(closes issue #17892)
Reported by: wdoekes
Patches:
bug17892-1.patch uploaded by jpeeler (license 325)
Tested by: wdoekes
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Stefan Schmidt [Fri, 1 Oct 2010 10:04:31 +0000 (10:04 +0000)]
don't iterate through all dialogs to find and delete old subscribes
On every incoming subscribe there is a iteration through all dialogs to find old subscribes and delete them. This is slow and not RFC conform. This was only needed in 1.2 cause a subscribe was not deleted when a dialog was destroyed, after 1.4 a subscribe get removed when its dialog is destroyed.
Review: https://reviewboard.asterisk.org/r/901/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@289623
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Tilghman Lesher [Thu, 30 Sep 2010 20:40:08 +0000 (20:40 +0000)]
Merged revisions 289543,289581 via svnmerge from
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r289543 | tilghman | 2010-09-30 12:50:52 -0500 (Thu, 30 Sep 2010) | 2 lines
More Solaris compatibility fixes
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r289581 | tilghman | 2010-09-30 15:23:10 -0500 (Thu, 30 Sep 2010) | 2 lines
Solaris fixes.
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Matthew Nicholson [Thu, 30 Sep 2010 19:54:59 +0000 (19:54 +0000)]
Merged revisions 289554 via svnmerge from
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r289554 | mnicholson | 2010-09-30 14:53:10 -0500 (Thu, 30 Sep 2010) | 11 lines
Merged revisions 289553 via svnmerge from
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r289553 | mnicholson | 2010-09-30 14:51:27 -0500 (Thu, 30 Sep 2010) | 4 lines
Properly handle channel allocation failures duing invites with replaces.
ABE-2588
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Richard Mudgett [Thu, 30 Sep 2010 19:35:47 +0000 (19:35 +0000)]
Merged revisions 289549 via svnmerge from
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r289549 | rmudgett | 2010-09-30 14:28:36 -0500 (Thu, 30 Sep 2010) | 17 lines
Merged revision 289547 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
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r289547 | rmudgett | 2010-09-30 14:16:36 -0500 (Thu, 30 Sep 2010) | 10 lines
In chan_misdn, the DivertingLegInformation2 DivertingNr is garbage when the number is restricted.
The same thing happens with DivertingLegInformation1 DivertedTo number.
The misdn_PresentedNumberUnscreened_extract() extracted the Unscreened
PartyNumber field unconditionally. It now checks the presented number
unscreened type to see if the PartyNumber was even present.
JIRA ABE-2595
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Russell Bryant [Thu, 30 Sep 2010 15:40:10 +0000 (15:40 +0000)]
Merged revisions 289426 via svnmerge from
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r289426 | russell | 2010-09-30 10:39:45 -0500 (Thu, 30 Sep 2010) | 22 lines
Merged revisions 289425 via svnmerge from
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r289425 | russell | 2010-09-30 10:37:29 -0500 (Thu, 30 Sep 2010) | 15 lines
Merged revisions 289424 via svnmerge from
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r289424 | russell | 2010-09-30 10:34:29 -0500 (Thu, 30 Sep 2010) | 8 lines
Fix a crash in app_sms.
Since the data being passed to the generator callback is on the stack of the
SMS() application, we must ensure that the generator is stopped before the
application exits.
ABE-2587
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Jason Parker [Wed, 29 Sep 2010 21:19:46 +0000 (21:19 +0000)]
Merged revisions 289340 via svnmerge from
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r289340 | qwell | 2010-09-29 16:12:43 -0500 (Wed, 29 Sep 2010) | 22 lines
Merged revisions 289339 via svnmerge from
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r289339 | qwell | 2010-09-29 16:03:47 -0500 (Wed, 29 Sep 2010) | 15 lines
Merged revisions 289338 via svnmerge from
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r289338 | qwell | 2010-09-29 15:56:26 -0500 (Wed, 29 Sep 2010) | 8 lines
Allow a manager originate to succeed on forwarded devices.
The timeout to wait for an answer was being set to 0 when a device forwarded to another
extension. We don't always need the timeout set like this, so make it an optional
parameter, and don't use it in this case.
ABE-2544
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Leif Madsen [Wed, 29 Sep 2010 20:29:11 +0000 (20:29 +0000)]
Merged revisions 289336 via svnmerge from
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r289336 | lmadsen | 2010-09-29 15:27:25 -0500 (Wed, 29 Sep 2010) | 9 lines
Merged revisions 289334 via svnmerge from
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r289334 | lmadsen | 2010-09-29 15:24:47 -0500 (Wed, 29 Sep 2010) | 1 line
Update sample documentation to note md5secret requirements.
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Russell Bryant [Wed, 29 Sep 2010 20:24:49 +0000 (20:24 +0000)]
Merged revisions 289333 via svnmerge from
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r289333 | russell | 2010-09-29 15:20:23 -0500 (Wed, 29 Sep 2010) | 11 lines
Merged revisions 289332 via svnmerge from
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r289332 | russell | 2010-09-29 15:15:57 -0500 (Wed, 29 Sep 2010) | 4 lines
Don't completely ignore md5secret from LDAP if the value does not begin with {md5}.
This fixes a problem that lmadsen ran in to where md5secret was not working for him.
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Matthew Nicholson [Wed, 29 Sep 2010 17:54:49 +0000 (17:54 +0000)]
Merged revisions 289300 via svnmerge from
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r289300 | mnicholson | 2010-09-29 12:53:54 -0500 (Wed, 29 Sep 2010) | 2 lines
Add 'ecm' to the sample fax config file
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Matthew Nicholson [Wed, 29 Sep 2010 17:08:56 +0000 (17:08 +0000)]
Merged revisions 289268 via svnmerge from
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r289268 | mnicholson | 2010-09-29 12:08:20 -0500 (Wed, 29 Sep 2010) | 5 lines
Update the CDR record when ast_channel_set_caller_event() is called
(related to issue #17569)
Reported by: tbelder
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Richard Mudgett [Wed, 29 Sep 2010 16:17:27 +0000 (16:17 +0000)]
Merged revisions 289253 via svnmerge from
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r289253 | rmudgett | 2010-09-29 11:16:47 -0500 (Wed, 29 Sep 2010) | 1 line
Make development error message indicate which channel.
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Matthew Nicholson [Wed, 29 Sep 2010 15:07:57 +0000 (15:07 +0000)]
Merged revisions 289179 via svnmerge from
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r289179 | mnicholson | 2010-09-29 10:04:56 -0500 (Wed, 29 Sep 2010) | 22 lines
Merged revisions 289178 via svnmerge from
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r289178 | mnicholson | 2010-09-29 10:04:11 -0500 (Wed, 29 Sep 2010) | 15 lines
Merged revisions 289177 via svnmerge from
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r289177 | mnicholson | 2010-09-29 10:03:27 -0500 (Wed, 29 Sep 2010) | 8 lines
Set the caller id on CDRs when it is set on the parent channel.
(closes issue #17569)
Reported by: tbelder
Patches:
17569.diff uploaded by tbelder (license 618)
Tested by: tbelder
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Brett Bryant [Tue, 28 Sep 2010 18:24:11 +0000 (18:24 +0000)]
Merged revisions 289099 via svnmerge from
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r289099 | bbryant | 2010-09-28 14:18:02 -0400 (Tue, 28 Sep 2010) | 28 lines
Merged revisions 289095 via svnmerge from
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r289095 | bbryant | 2010-09-28 14:14:19 -0400 (Tue, 28 Sep 2010) | 21 lines
Merged revisions 289094 via svnmerge from
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r289094 | bbryant | 2010-09-28 14:10:19 -0400 (Tue, 28 Sep 2010) | 14 lines
Fixes an issue with the Newchannel AMI event during the Masquerading process.
Fixes an issue with the Newchannel AMI event during the Masquerading process,
where no Newchannel AMI event was generated for the psuedo channel used during
the masquerading process.
(closes issue #17987)
Reported by: RadicAlish
Patches:
newchannel.patch.txt uploaded by RadicAlish (license 1122)
Tested by: RadicAlish
Review: https://reviewboard.asterisk.org/r/937/
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Tilghman Lesher [Tue, 28 Sep 2010 18:20:20 +0000 (18:20 +0000)]
Merged revisions 289104 via svnmerge from
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r289104 | tilghman | 2010-09-28 13:18:43 -0500 (Tue, 28 Sep 2010) | 4 lines
Solaris compatibility fixes
Review: https://reviewboard.asterisk.org/r/942/
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Richard Mudgett [Tue, 28 Sep 2010 01:10:25 +0000 (01:10 +0000)]
Merged revisions 289057 via svnmerge from
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r289057 | rmudgett | 2010-09-27 20:04:37 -0500 (Mon, 27 Sep 2010) | 5 lines
Avoid deadlock processing incoming AOC-E messages.
Deadlock avoidance for the owner channel was not done when processing
incoming AOC-E messages.
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Richard Mudgett [Tue, 28 Sep 2010 00:36:27 +0000 (00:36 +0000)]
Merged revisions 289054-289055 via svnmerge from
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r289054 | rmudgett | 2010-09-27 19:32:18 -0500 (Mon, 27 Sep 2010) | 1 line
Break up long ast_manager_event_multichan() event lines.
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r289055 | rmudgett | 2010-09-27 19:35:25 -0500 (Mon, 27 Sep 2010) | 1 line
Revert stuff not ready for commit in -r289054.
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David Vossel [Mon, 27 Sep 2010 22:03:54 +0000 (22:03 +0000)]
For an INVITE transaction, treat all 2XX responses the same as a 200.
ABE-2305
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Olle Johansson [Mon, 27 Sep 2010 19:45:56 +0000 (19:45 +0000)]
Formatting fixes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@288993
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Olle Johansson [Mon, 27 Sep 2010 19:30:18 +0000 (19:30 +0000)]
Formating changes
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Tilghman Lesher [Mon, 27 Sep 2010 18:39:05 +0000 (18:39 +0000)]
Merged revisions 288961 via svnmerge from
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r288961 | tilghman | 2010-09-27 13:37:41 -0500 (Mon, 27 Sep 2010) | 5 lines
Still build SIP, even if res_crypto cannot be built (use, not depend).
(closes issue #18062)
Reported by: a user on the mailing list
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Russell Bryant [Mon, 27 Sep 2010 13:04:23 +0000 (13:04 +0000)]
Merged revisions 288927 via svnmerge from
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r288927 | russell | 2010-09-27 08:03:43 -0500 (Mon, 27 Sep 2010) | 2 lines
Fix some documentation typos and spelling errors.
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Russell Bryant [Mon, 27 Sep 2010 12:42:30 +0000 (12:42 +0000)]
Merged revisions 288925 via svnmerge from
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r288925 | russell | 2010-09-27 07:42:10 -0500 (Mon, 27 Sep 2010) | 2 lines
Fix a documentation spelling error.
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Alexandr Anikin [Sat, 25 Sep 2010 07:58:23 +0000 (07:58 +0000)]
small correction for verbose print h.323 packets
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David Vossel [Fri, 24 Sep 2010 17:59:47 +0000 (17:59 +0000)]
Merged revisions 288852 via svnmerge from
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r288852 | dvossel | 2010-09-24 12:58:57 -0500 (Fri, 24 Sep 2010) | 5 lines
Append Retry-After header on 500 error response to Re-INVITE according to RFC3261 section 14.2.
ABE-2301
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David Vossel [Fri, 24 Sep 2010 17:06:02 +0000 (17:06 +0000)]
Merged revisions 288821 via svnmerge from
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r288821 | dvossel | 2010-09-24 12:05:12 -0500 (Fri, 24 Sep 2010) | 4 lines
Inspect Require header on BYE transaction according to RFC3261 section 8.2.2.3.
ABE-2293
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Terry Wilson [Fri, 24 Sep 2010 16:11:19 +0000 (16:11 +0000)]
Merged revisions 288748 via svnmerge from
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r288748 | twilson | 2010-09-24 09:02:27 -0700 (Fri, 24 Sep 2010) | 19 lines
Merged revisions 288747 via svnmerge from
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r288747 | twilson | 2010-09-24 08:37:39 -0700 (Fri, 24 Sep 2010) | 12 lines
Merged revisions 288746 via svnmerge from
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r288746 | twilson | 2010-09-24 08:26:09 -0700 (Fri, 24 Sep 2010) | 5 lines
Don't fail a masquerade if it is already being hung up
This avoids noise on some Local channel situations where we don't use /n.
Thanks to Alec Davis for the suggestion.
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Tilghman Lesher [Fri, 24 Sep 2010 13:55:11 +0000 (13:55 +0000)]
Merged revisions 288713 via svnmerge from
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r288713 | tilghman | 2010-09-24 08:54:17 -0500 (Fri, 24 Sep 2010) | 12 lines
Merged revisions 288712 via svnmerge from
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r288712 | tilghman | 2010-09-24 08:53:30 -0500 (Fri, 24 Sep 2010) | 5 lines
Solaris won't printf a NULL.
(closes issue #18041)
Reported by: asgaroth
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Tilghman Lesher [Fri, 24 Sep 2010 03:43:14 +0000 (03:43 +0000)]
Merged revisions 288640 via svnmerge from
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r288640 | tilghman | 2010-09-23 22:42:37 -0500 (Thu, 23 Sep 2010) | 2 lines
Export timersub for platforms which do not have it
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Tilghman Lesher [Fri, 24 Sep 2010 03:41:02 +0000 (03:41 +0000)]
Merged revisions 288638 via svnmerge from
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r288638 | tilghman | 2010-09-23 22:39:29 -0500 (Thu, 23 Sep 2010) | 16 lines
Merged revisions 288637 via svnmerge from
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r288637 | tilghman | 2010-09-23 22:36:01 -0500 (Thu, 23 Sep 2010) | 9 lines
Merged revisions 288636 via svnmerge from
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r288636 | tilghman | 2010-09-23 22:20:24 -0500 (Thu, 23 Sep 2010) | 2 lines
Solaris compatibility fixes
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Tilghman Lesher [Thu, 23 Sep 2010 18:45:41 +0000 (18:45 +0000)]
Merged revisions 288606 via svnmerge from
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r288606 | tilghman | 2010-09-23 13:44:44 -0500 (Thu, 23 Sep 2010) | 2 lines
Add note about the checkhangup option of ${CHANNEL()}
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Terry Wilson [Thu, 23 Sep 2010 18:08:23 +0000 (18:08 +0000)]
Merged revisions 288572 via svnmerge from
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r288572 | twilson | 2010-09-23 13:05:16 -0500 (Thu, 23 Sep 2010) | 2 lines
Make AMI honor enabled=no
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Terry Wilson [Wed, 22 Sep 2010 23:20:27 +0000 (23:20 +0000)]
Merged revisions 288507 via svnmerge from
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r288507 | twilson | 2010-09-22 16:18:27 -0700 (Wed, 22 Sep 2010) | 22 lines
Merged revisions 288500 via svnmerge from
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r288500 | twilson | 2010-09-22 16:10:09 -0700 (Wed, 22 Sep 2010) | 15 lines
Merged revisions 288499 via svnmerge from
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r288499 | twilson | 2010-09-22 16:00:30 -0700 (Wed, 22 Sep 2010) | 8 lines
Don't let a Local channel get bridged to itself
If a local channel gets bridged to itself, it becomes orphaned with no devices
left to actually tell it to hang up. This patch modifies local_fixup() to detect
this case and deny it.
Review: https://reviewboard.asterisk.org/r/934
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David Vossel [Wed, 22 Sep 2010 17:50:32 +0000 (17:50 +0000)]
Merged revisions 288418 via svnmerge from
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r288418 | dvossel | 2010-09-22 12:49:56 -0500 (Wed, 22 Sep 2010) | 18 lines
Merged revisions 288417 via svnmerge from
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r288417 | dvossel | 2010-09-22 12:49:05 -0500 (Wed, 22 Sep 2010) | 11 lines
Merged revisions 288416 via svnmerge from
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r288416 | dvossel | 2010-09-22 12:48:15 -0500 (Wed, 22 Sep 2010) | 5 lines
RFC3261 section 12.2 explicitly says out of order requests are responded with a 500 Server Internal Error response.
ABE-2458
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David Vossel [Wed, 22 Sep 2010 17:13:05 +0000 (17:13 +0000)]
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r288345 | dvossel | 2010-09-22 11:59:14 -0500 (Wed, 22 Sep 2010) | 16 lines
Merged revisions 288344 via svnmerge from
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r288344 | dvossel | 2010-09-22 11:53:28 -0500 (Wed, 22 Sep 2010) | 9 lines
Merged revisions 288343 via svnmerge from
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r288343 | dvossel | 2010-09-22 11:49:56 -0500 (Wed, 22 Sep 2010) | 2 lines
During check_pendings, if the dialog is terminated with a CANCEL, change the invitestate to INV_CANCEL like in sip_hangup.
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Russell Bryant [Wed, 22 Sep 2010 16:46:20 +0000 (16:46 +0000)]
Merged revisions 288341 via svnmerge from
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r288341 | russell | 2010-09-22 11:45:18 -0500 (Wed, 22 Sep 2010) | 25 lines
Merged revisions 288340 via svnmerge from
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r288340 | russell | 2010-09-22 11:44:13 -0500 (Wed, 22 Sep 2010) | 18 lines
Merged revisions 288339 via svnmerge from
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r288339 | russell | 2010-09-22 11:39:16 -0500 (Wed, 22 Sep 2010) | 11 lines
Fix a 100% CPU consumption problem when setting console=yes in asterisk.conf.
The handling of -c and console=yes should be the same, but they were not.
When you specify -c, it sets both a flag for console module and for asterisk
not to fork() off into the background. The handling of console=yes only set
console mode, so you would end up with a background process() trying to run
the Asterisk console and freaking out since it didn't have anything to read
input from.
Thanks to beagles for reporting and helping debug the problem!
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Tilghman Lesher [Wed, 22 Sep 2010 15:18:49 +0000 (15:18 +0000)]
Merged revisions 288268 via svnmerge from
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r288268 | tilghman | 2010-09-22 10:14:02 -0500 (Wed, 22 Sep 2010) | 30 lines
Merged revisions 288267 via svnmerge from
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r288267 | tilghman | 2010-09-22 10:11:09 -0500 (Wed, 22 Sep 2010) | 23 lines
Merged revisions 288265-288266 via svnmerge from
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r288265 | tilghman | 2010-09-22 09:48:04 -0500 (Wed, 22 Sep 2010) | 9 lines
Allow the encoding to be set, in case local charset does not agree with database.
(closes issue #16940)
Reported by: jamicque
Patches:
20100827__issue16940.diff.txt uploaded by tilghman (license 14)
20100921__issue16940__1.6.2.diff.txt uploaded by tilghman (license 14)
Tested by: jamicque
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r288266 | tilghman | 2010-09-22 10:04:52 -0500 (Wed, 22 Sep 2010) | 5 lines
Document addition of encoding parameter.
(issue #16940)
Reported by: jamicque
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Richard Mudgett [Wed, 22 Sep 2010 00:08:49 +0000 (00:08 +0000)]
Merged revisions 288194 via svnmerge from
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r288194 | rmudgett | 2010-09-21 19:06:21 -0500 (Tue, 21 Sep 2010) | 40 lines
Merged revisions 288193 via svnmerge from
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r288193 | rmudgett | 2010-09-21 19:03:37 -0500 (Tue, 21 Sep 2010) | 33 lines
Merged revisions 288192 via svnmerge from
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r288192 | rmudgett | 2010-09-21 18:55:58 -0500 (Tue, 21 Sep 2010) | 26 lines
In chan_iax2.c:schedule_delivery() calls ast_bridged_channel() on an unlocked channel.
Near the beginning of schedule_delivery(), ast_bridged_channel() is called
on iaxs[fr->callno]->owner. However, the channel is not locked, which can
result in ast_bridged_channel() crashing should owner->tech change to a
technology that doesn't implement bridged_channel.
I also fixed the other calls to ast_bridged_channel() in chan_iax2.c since
the owner lock was not held there either.
Converted the existing channel deadlock avoidance to use
iax2_lock_owner(). Using the new function simplified some awkward code.
In the process of fixing the locking on ast_bridged_channel(), I also
found a memory leak in socket_process() for v1.6.2 and v1.8. The local
struct variable ies.vars is not freed on early/abnormal function exits.
(closes issue #17919)
Reported by: rain
Patches:
issue17919_v1.4.patch uploaded by rmudgett (license 664)
issue17919_w_leak_v1.6.2.patch uploaded by rmudgett (license 664)
issue17919_w_leak_v1.8.patch uploaded by rmudgett (license 664)
Review: https://reviewboard.asterisk.org/r/926/
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Tilghman Lesher [Tue, 21 Sep 2010 22:58:10 +0000 (22:58 +0000)]
Merged revisions 288159 via svnmerge from
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r288159 | tilghman | 2010-09-21 17:57:22 -0500 (Tue, 21 Sep 2010) | 29 lines
Merged revisions 288113 via svnmerge from
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r288113 | tilghman | 2010-09-21 16:59:46 -0500 (Tue, 21 Sep 2010) | 22 lines
Merged revisions 288112 via svnmerge from
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r288112 | tilghman | 2010-09-21 16:58:13 -0500 (Tue, 21 Sep 2010) | 15 lines
Try both the encoded and unencoded subscription URI for a match in hints.
When a phone sends an encoded URI for a subscription, the URI is not matched
with the actual hint that is in decoded format. For example, if we have an
extension with a hint that is named: "#5601" or "*5601", the subscription will
work fine if the phone subscribes with an already decoded URI, but when it's
decoded like "%255601" or "%2A5601", Asterisk is unable to match it with the
correct hint.
(closes issue #17785)
Reported by: ramonpeek
Patches:
20100831__issue17785.diff.txt uploaded by tilghman (license 14)
Tested by: ramonpeek
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Paul Belanger [Tue, 21 Sep 2010 22:28:21 +0000 (22:28 +0000)]
Merged revisions 288157 via svnmerge from
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r288157 | pabelanger | 2010-09-21 18:26:15 -0400 (Tue, 21 Sep 2010) | 15 lines
Merged revisions 288147 via svnmerge from
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r288147 | pabelanger | 2010-09-21 18:22:43 -0400 (Tue, 21 Sep 2010) | 9 lines
Setup timer before set_config().
(closes issue #18019)
Reported by: Netview
Patches:
issue_0018019.patch uploaded by pabelanger (license 224)
Tested by: Netview
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Richard Mudgett [Tue, 21 Sep 2010 21:04:04 +0000 (21:04 +0000)]
Merged revisions 288082 via svnmerge from
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r288082 | rmudgett | 2010-09-21 16:03:28 -0500 (Tue, 21 Sep 2010) | 1 line
Add note in party manipulation chapter on interception macros.
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Richard Mudgett [Tue, 21 Sep 2010 20:33:20 +0000 (20:33 +0000)]
Merged revisions 288079-288080 via svnmerge from
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r288079 | rmudgett | 2010-09-21 15:29:51 -0500 (Tue, 21 Sep 2010) | 2 lines
Protect channel access in CONNECTED_LINE and REDIRECTING interception macro launch code.
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r288080 | rmudgett | 2010-09-21 15:29:59 -0500 (Tue, 21 Sep 2010) | 8 lines
Simplify locking code for REDIRECTING interception macro when forwarding a call.
Simplified the locking code by using a local copy of the redirecting party
information in app_dial.c:do_forward() and app_queue.c:wait_for_answer()
for launching the REDIRECTING interception macro when a call is forwarded.
Reduced the lock time of the 'o->chan' and 'in' channels.
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Stefan Schmidt [Tue, 21 Sep 2010 20:27:04 +0000 (20:27 +0000)]
Instead of iterate through all dialogs, add two separte container for needdestroy and rtptimeout
adding two dialog container, one for dialogs which need destroy, another for rtptimeout checks.
both container will be checked on every loop of do_monitor instead of iterate through all dialogs.
(closes issue #17912)
Reported by: schmidts
Tested by: schmidts
Review: https://reviewboard.asterisk.org/r/917/
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Brett Bryant [Tue, 21 Sep 2010 19:50:46 +0000 (19:50 +0000)]
Merged revisions 288007 via svnmerge from
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r288007 | bbryant | 2010-09-21 15:48:53 -0400 (Tue, 21 Sep 2010) | 21 lines
Merged revisions 288006 via svnmerge from
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r288006 | bbryant | 2010-09-21 15:46:20 -0400 (Tue, 21 Sep 2010) | 14 lines
Merged revisions 288005 via svnmerge from
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r288005 | bbryant | 2010-09-21 15:43:46 -0400 (Tue, 21 Sep 2010) | 8 lines
Add a check to fix a rare segmentation fault you'd get if ast_frdup couldn't allocate
memory on the first frame being queued in ast_queue_frame.
(closes issue #17882)
Reported by: seanbright
Tested by: seanbright
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Tilghman Lesher [Tue, 21 Sep 2010 19:09:15 +0000 (19:09 +0000)]
Merged revisions 287935 via svnmerge from
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r287935 | tilghman | 2010-09-21 14:08:36 -0500 (Tue, 21 Sep 2010) | 16 lines
Merged revisions 287934 via svnmerge from
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r287934 | tilghman | 2010-09-21 14:07:53 -0500 (Tue, 21 Sep 2010) | 9 lines
Merged revisions 287933 via svnmerge from
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r287933 | tilghman | 2010-09-21 14:07:07 -0500 (Tue, 21 Sep 2010) | 2 lines
Less than zero is an error, not any non-zero value.
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Terry Wilson [Tue, 21 Sep 2010 19:04:57 +0000 (19:04 +0000)]
Merged revisions 287931 via svnmerge from
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r287931 | twilson | 2010-09-21 14:02:40 -0500 (Tue, 21 Sep 2010) | 2 lines
Revert change in favor of a more targeted fix
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David Vossel [Tue, 21 Sep 2010 18:33:18 +0000 (18:33 +0000)]
Merged revisions 287929 via svnmerge from
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r287929 | dvossel | 2010-09-21 13:32:12 -0500 (Tue, 21 Sep 2010) | 4 lines
Send a "415 Unsupported Media Type" after failure to process sdp due to unknown Content-Encoding header.
ABE-2258
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Richard Mudgett [Tue, 21 Sep 2010 15:54:12 +0000 (15:54 +0000)]
Merged revisions 287897 via svnmerge from
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r287897 | rmudgett | 2010-09-21 10:53:19 -0500 (Tue, 21 Sep 2010) | 1 line
Cut-n-paste error in builtin_blindtransfer().
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Russell Bryant [Tue, 21 Sep 2010 15:45:46 +0000 (15:45 +0000)]
Merged revisions 287895 via svnmerge from
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r287895 | russell | 2010-09-21 10:43:33 -0500 (Tue, 21 Sep 2010) | 10 lines
Don't use ast_strdupa() from within the arguments to a function.
(closes issue #17902)
Reported by: afried
Patches:
issue_17902.rev1.txt uploaded by russell (license 2)
Tested by: russell
Review: https://reviewboard.asterisk.org/r/927/
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Tilghman Lesher [Tue, 21 Sep 2010 15:27:10 +0000 (15:27 +0000)]
Merged revisions 287893 via svnmerge from
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r287893 | tilghman | 2010-09-21 10:24:47 -0500 (Tue, 21 Sep 2010) | 9 lines
Anonymous callerid needs a "sip:" uri prefix.
(closes issue #17981)
Reported by: avalentin
Patches:
sip-anonymous-aastra.patch uploaded by avalentin (license 1107)
(plus an additional fix by me)
Tested by: avalentin
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Russell Bryant [Tue, 21 Sep 2010 13:45:30 +0000 (13:45 +0000)]
Merged revisions 287863 via svnmerge from
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r287863 | russell | 2010-09-21 08:41:41 -0500 (Tue, 21 Sep 2010) | 2 lines
Fix a regression in verbose logger processing.
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Terry Wilson [Tue, 21 Sep 2010 04:39:30 +0000 (04:39 +0000)]
Merged revisions 287833 via svnmerge from
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r287833 | twilson | 2010-09-20 23:37:44 -0500 (Mon, 20 Sep 2010) | 3 lines
Don't generate connected line buffer twice for comparison
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Terry Wilson [Tue, 21 Sep 2010 00:11:59 +0000 (00:11 +0000)]
Merged revisions 287757 via svnmerge from
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r287757 | twilson | 2010-09-20 18:51:38 -0500 (Mon, 20 Sep 2010) | 7 lines
Avoid infinite loop with certain local channel connected line updates
Compare connected line data before sending a connected line indication to avoid
possible loops.
Review: https://reviewboard.asterisk.org/r/932/
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Brett Bryant [Tue, 21 Sep 2010 00:04:54 +0000 (00:04 +0000)]
Merged revisions 287760 via svnmerge from
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r287760 | bbryant | 2010-09-20 20:00:23 -0400 (Mon, 20 Sep 2010) | 30 lines
Merged revisions 287759 via svnmerge from
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r287759 | bbryant | 2010-09-20 19:58:26 -0400 (Mon, 20 Sep 2010) | 23 lines
Merged revisions 287758 via svnmerge from
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r287758 | bbryant | 2010-09-20 19:57:08 -0400 (Mon, 20 Sep 2010) | 16 lines
Fix misvalidation of meetme pins in conjunction with the 'a' MeetMe flag.
When using the 'a' MeetMe flag and having a user and admin pin setup for your
conference, using the user pin would gain you admin priviledges. Also, when no
user pin was set, an admin pin was, the 'a' MeetMe flag wasn't used, and the
user tried to enter a conference then they were still prompted for a pin and
forced to hit #.
(closes issue #17908)
Reported by: kuj
Patches:
pins_2.patch uploaded by kuj (license 1111)
Tested by: kuj
Review: [full review board URL with trailing slash]
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Terry Wilson [Tue, 21 Sep 2010 00:01:33 +0000 (00:01 +0000)]
Add alecdavis' commit to merged props
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Terry Wilson [Tue, 21 Sep 2010 00:00:41 +0000 (00:00 +0000)]
Add merge properties back.
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Alec L Davis [Mon, 20 Sep 2010 23:42:56 +0000 (23:42 +0000)]
Merged revisions 287685 via svnmerge from
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r287685 | alecdavis | 2010-09-21 11:16:45 +1200 (Tue, 21 Sep 2010) | 18 lines
ast_channel_masquerade: Avoid recursive masquerades.
Check all 4 combinations of (original/clonechan) * (masq/masqr).
Initially original->masq and clonechan->masqr were only checked.
It's possible with multiple masq's planned - and not yet executed, that
the 'original' chan could already have another masq'd into it - thus original->masqr
would be set, that masqr would lost.
Likewise for the clonechan->masq.
(closes issue #16057;#17363)
Reported by: amorsen;davidw,alecdavis
Patches:
based on bug16057.diff4.txt uploaded by alecdavis (license 585)
Tested by: ramonpeek, davidw, alecdavis
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Richard Mudgett [Mon, 20 Sep 2010 23:18:41 +0000 (23:18 +0000)]
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r287683 | rmudgett | 2010-09-20 18:14:42 -0500 (Mon, 20 Sep 2010) | 9 lines
The inalarm flag was not set in sig_analog struct if the port is initially in alarm.
Fixed initial inalarm value for sig_analog ports.
Along with -r261007, this gets the inalarm flag in sync with chan_dahdi
for sig_analog ports.
(closes issue #16983)
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Alec L Davis [Mon, 20 Sep 2010 22:24:51 +0000 (22:24 +0000)]
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r287661 | alecdavis | 2010-09-21 10:21:50 +1200 (Tue, 21 Sep 2010) | 14 lines
ast_do_masquerade. Keep channels ao2_container locked while unlink and linking channels.
Previously, Masquerade would unlock 'original' and 'clonechan' and allow another masq thread to run.
End result would be corrupted memory, and the frequent report 'Bad Magic Number'.
(closes issue #17801,#17710)
Reported by: notthematrix
Patches:
Based on bug17801.diff1.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
Review: https://reviewboard.asterisk.org/r/928
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David Vossel [Mon, 20 Sep 2010 22:16:37 +0000 (22:16 +0000)]
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r287647 | dvossel | 2010-09-20 17:09:16 -0500 (Mon, 20 Sep 2010) | 21 lines
Addition of the FrameHook API (AKA AwesomeHooks)
So far all our tools for viewing and manipulating media streams
within Asterisk have been entirely focused on audio. That made
sense then, but is not scalable now. The FrameHook API lets us
tap into and manipulate _ANY_ type of media or signaling passed
on a channel present today or in the future. This tool is a step
in the direction of expanding Asterisk's boundaries and will help
generate some rather interesting applications in the future.
In addition to the FrameHook API, a simple dialplan function
exercising the api has been included as well. This function
is called FRAME_TRACE(). FRAME_TRACE() allows for the internal
ast_frames read and written to a channel to be output. Filters
can be placed on this function to debug only certain types of frames.
This function could be thought of as an internal way of doing
ast_frame packet captures.
Review: https://reviewboard.asterisk.org/r/925/
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David Vossel [Mon, 20 Sep 2010 21:35:46 +0000 (21:35 +0000)]
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r287645 | dvossel | 2010-09-20 16:34:15 -0500 (Mon, 20 Sep 2010) | 9 lines
Fixes issue with registrations not working properly with pedantic=yes.
(closes issue #18017)
Reported by: schmidts
Patches:
issues_18017_v1.diff uploaded by dvossel (license 671)
Tested by: schmidts
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Jason Parker [Mon, 20 Sep 2010 21:30:12 +0000 (21:30 +0000)]
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r287643 | qwell | 2010-09-20 16:29:46 -0500 (Mon, 20 Sep 2010) | 15 lines
Merged revisions 287642 via svnmerge from
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r287642 | qwell | 2010-09-20 16:28:32 -0500 (Mon, 20 Sep 2010) | 8 lines
Don't crash when parking a non-bridged call.
(closes issue #17680)
Reported by: jmhunter
Patches:
chan_skinny-park-v1.txt uploaded by DEA (license 3)
Tested by: jmhunter, DEA
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Brett Bryant [Mon, 20 Sep 2010 21:25:01 +0000 (21:25 +0000)]
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r287639 | bbryant | 2010-09-20 17:19:12 -0400 (Mon, 20 Sep 2010) | 8 lines
Fixes an error with the logger that caused verbose messages to be spammed to the
screen if syslog was configured in logger.conf
(closes issue #17974)
Reported by: lmadsen
Review: https://reviewboard.asterisk.org/r/915/
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Matthew Nicholson [Mon, 20 Sep 2010 15:57:52 +0000 (15:57 +0000)]
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r287559 | mnicholson | 2010-09-20 10:57:14 -0500 (Mon, 20 Sep 2010) | 21 lines
Merged revisions 287558 via svnmerge from
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r287558 | mnicholson | 2010-09-20 10:56:21 -0500 (Mon, 20 Sep 2010) | 14 lines
Use ast_str when processing hint state changes
Merged revisions 287555 via svnmerge from
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r287555 | mnicholson | 2010-09-20 10:48:14 -0500 (Mon, 20 Sep 2010) | 5 lines
Use ast_dynamic_str when processing hint state changes
(related to issue #17928)
Reported by: mdu113
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Olle Johansson [Sun, 19 Sep 2010 16:12:08 +0000 (16:12 +0000)]
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r287471 | oej | 2010-09-19 18:09:28 +0200 (Sön, 19 Sep 2010) | 21 lines
Merged revisions 287470 via svnmerge from
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r287470 | oej | 2010-09-19 18:06:10 +0200 (Sön, 19 Sep 2010) | 14 lines
Merged revisions 287469 via svnmerge from
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r287469 | oej | 2010-09-19 17:56:50 +0200 (Sön, 19 Sep 2010) | 7 lines
Make sure we always free variables properly in manager originate.
(closes issue #17891)
reported, solved and tested by oej
Review: https://reviewboard.asterisk.org/r/869/
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Tilghman Lesher [Fri, 17 Sep 2010 21:10:02 +0000 (21:10 +0000)]
Merged revisions 287388 via svnmerge from
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r287388 | tilghman | 2010-09-17 16:08:54 -0500 (Fri, 17 Sep 2010) | 21 lines
Merged revisions 287387 via svnmerge from
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r287387 | tilghman | 2010-09-17 16:08:00 -0500 (Fri, 17 Sep 2010) | 14 lines
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r287386 | tilghman | 2010-09-17 16:06:03 -0500 (Fri, 17 Sep 2010) | 7 lines
Blank columns should get set on reload, not ignored.
(closes issue #16893)
Reported by: haakon
Patches:
20100818__issue16893.diff.txt uploaded by tilghman (license 14)
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Matthew Nicholson [Fri, 17 Sep 2010 13:38:22 +0000 (13:38 +0000)]
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r287309 | mnicholson | 2010-09-17 08:37:10 -0500 (Fri, 17 Sep 2010) | 19 lines
Merged revisions 287308 via svnmerge from
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r287308 | mnicholson | 2010-09-17 08:36:07 -0500 (Fri, 17 Sep 2010) | 12 lines
Merged revisions 287307 via svnmerge from
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r287307 | mnicholson | 2010-09-17 08:34:34 -0500 (Fri, 17 Sep 2010) | 5 lines
Use ast_strdup() instead of ast_strdupa() while processing in ast_hint_state_changed().
(related to issue #17928)
Reported by: mdu113
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Jan Kalab [Fri, 17 Sep 2010 08:46:45 +0000 (08:46 +0000)]
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r287269 | pitel | 2010-09-17 10:37:49 +0200 (Pá, 17 zář 2010) | 8 lines
Support for HTTP redirects in calendar's URL
libneon does not support HTTP redirects (3xx responses) by default. You must tell it to follow them.
Also, another little unsigned int fix.
(closes issue #17776)
Review: https://reviewboard.asterisk.org/r/921/
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r287270 | pitel | 2010-09-17 10:42:37 +0200 (Pá, 17 zář 2010) | 6 lines
Asterisk crashing because of double free when EWS request fails
The free is done later in code. I think ast_free() should have built in checks for double free.
(closes issue #17782)
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r287271 | pitel | 2010-09-17 10:44:28 +0200 (Pá, 17 zář 2010) | 6 lines
Events are visible after they were removed from EWS calendar
Because we must merge calendar even when it's empty.
(closes issue #17786)
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Jason Parker [Thu, 16 Sep 2010 22:05:14 +0000 (22:05 +0000)]
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r287195 | qwell | 2010-09-16 17:04:38 -0500 (Thu, 16 Sep 2010) | 7 lines
Don't fail when running the Debian init script directly (as one would normally do).
readlink apparently returns 1 when the arg isn't a symlink, which caused the script to exit.
(closes issue #17910)
Reported by: wurstsalat
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Russell Bryant [Thu, 16 Sep 2010 22:00:15 +0000 (22:00 +0000)]
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r287193 | russell | 2010-09-16 16:57:51 -0500 (Thu, 16 Sep 2010) | 4 lines
Set the default for "autofill" and "shared_lastcall" to "yes" in queues.conf.
Review: https://reviewboard.asterisk.org/r/922/
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Matthew Nicholson [Thu, 16 Sep 2010 20:08:51 +0000 (20:08 +0000)]
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r287120 | mnicholson | 2010-09-16 15:07:38 -0500 (Thu, 16 Sep 2010) | 22 lines
Merged revisions 287119 via svnmerge from
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r287119 | mnicholson | 2010-09-16 15:06:16 -0500 (Thu, 16 Sep 2010) | 15 lines
Merged revisions 287118 via svnmerge from
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r287118 | mnicholson | 2010-09-16 15:04:46 -0500 (Thu, 16 Sep 2010) | 8 lines
Don't limit hint processing in ast_hint_state_changed() to AST_MAX_EXTENSION length strings.
(closes issue #17928)
Reported by: mdu113
Patches:
20100831__issue17928.diff.txt uploaded by tilghman (license 14)
Tested by: mdu113
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Matthew Nicholson [Thu, 16 Sep 2010 19:55:21 +0000 (19:55 +0000)]
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r287116 | mnicholson | 2010-09-16 14:54:48 -0500 (Thu, 16 Sep 2010) | 22 lines
Merged revisions 287115 via svnmerge from
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r287115 | mnicholson | 2010-09-16 14:53:41 -0500 (Thu, 16 Sep 2010) | 15 lines
Merged revisions 287114 via svnmerge from
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r287114 | mnicholson | 2010-09-16 14:52:39 -0500 (Thu, 16 Sep 2010) | 8 lines
Don't stop printing cdr variables if we encounter one with a blank name or value.
(closes issue #17900)
Reported by: under
Patches:
core-show-channel-cdr-fix1.diff uploaded by mnicholson (license 96)
Tested by: mnicholson
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Olle Johansson [Thu, 16 Sep 2010 16:49:28 +0000 (16:49 +0000)]
We do not handle AST_CAUSE_INTERWORKING which we set on a lot of incoming
SIP messages. Adding error based on RFC 3398 recommendations.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287087
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Olle Johansson [Thu, 16 Sep 2010 16:48:08 +0000 (16:48 +0000)]
Add doxygen docs for indications.c
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287086
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Terry Wilson [Wed, 15 Sep 2010 22:28:29 +0000 (22:28 +0000)]
Merged revisions 287056 via svnmerge from
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r287056 | twilson | 2010-09-15 17:17:17 -0500 (Wed, 15 Sep 2010) | 10 lines
Don't hang up a call on an SRTP unprotect failure
Also make it more obvious when there is an issue en/decrypting.
(closes issue #17563)
Reported by: Alexcr
Patches:
res_srtp.c.patch uploaded by sfritsch (license 1089)
Tested by: twilson
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Jeff Peeler [Wed, 15 Sep 2010 21:00:03 +0000 (21:00 +0000)]
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r287020 | jpeeler | 2010-09-15 15:58:39 -0500 (Wed, 15 Sep 2010) | 1 line
fix uninintialized variable
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Richard Mudgett [Wed, 15 Sep 2010 20:56:21 +0000 (20:56 +0000)]
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r287017 | rmudgett | 2010-09-15 15:53:38 -0500 (Wed, 15 Sep 2010) | 65 lines
Merged revision 287014 from
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r287014 | rmudgett | 2010-09-15 15:32:24 -0500 (Wed, 15 Sep 2010) | 58 lines
The handling of call transfer signaling for mISDN PTMP is not fully implemented.
The handling of call transfer signaling for mISDN PTMP is not fully
implemented. The signaling of number updates with ISDN/DSS1 ECT
supplementary services (ETS 300 369-1) comes along with a notification
indicator IE and redirection number IE for PTMP. The implementation in
the current Asterisk mISDN channel unfortunately can handle these
information elements only in a NOTIFY message. These information elements
are also signaled in a FACILTY message with a RequestSubaddress facility,
when the subscriber is already in the active state (see 9.2.4 and 9.2.5 of
ETS 300 369-1).
**********
abe_2526_ast.patch
* Added support to handle the notification indicator IE and redirection
number IE with the RequestSubaddress facility.
* Made misdn_update_connected_line() send a NOTIFY message if Asterisk
originated the call and it is not connected yet.
* Made misdn_update_connected_line() send a FACILITY message if the call
is already connected.
This patch requires the presence of the associated mISDN patches to
compile. I had to enhance mISDN to allow the notification indicator IE
and the redirection number IE to be used with a FACILITY message. Earlier
versions of the Digium enhanced mISDN are no longer going to work.
**********
abe_2526_misdn.patch
* Made an incoming FACILITY message allow the presence of the notification
indicator IE and the redirection number IE.
**********
abe_2526_misdnuser_v3.patch
* Added support to send and receive a FACILITY message with the
notification indicator IE and the redirection number IE.
* Added the ability to send a NOTIFY message in PTMP/NT mode to all
responding subcalls in Q.931 states 6, 7, 8, 9, and 25.
**********
Patches:
abe_2526_ast.patch uploaded by rmudgett (license 664)
abe_2526_misdn.patch uploaded by rmudgett (license 664)
abe_2526_misdnuser_v3.patch uploaded by rmudgett (license 664)
Tested by: rmudgett and reporter
JIRA SWP-2146
JIRA ABE-2526
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Jeff Peeler [Wed, 15 Sep 2010 20:36:51 +0000 (20:36 +0000)]
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r287015 | jpeeler | 2010-09-15 15:32:52 -0500 (Wed, 15 Sep 2010) | 21 lines
Merged revisions 286998 via svnmerge from
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r286998 | jpeeler | 2010-09-15 15:28:02 -0500 (Wed, 15 Sep 2010) | 14 lines
Merged revisions 286941 via svnmerge from
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r286941 | jpeeler | 2010-09-15 15:08:52 -0500 (Wed, 15 Sep 2010) | 7 lines
Ensure mailbox is not filled to capacity before doing message forwarding.
Specifically, before prompting to record a prepended message the capacity is
checked first. If the mailbox is full the extension will be reprompted.
ABE-2517
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Jeff Peeler [Wed, 15 Sep 2010 19:23:56 +0000 (19:23 +0000)]
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r286931 | jpeeler | 2010-09-15 14:22:15 -0500 (Wed, 15 Sep 2010) | 16 lines
Add parking extension for non-default parking lots.
This is a new feature that allows for parking to custom parking lots to be
accessed directly, rather than with channel variables or by changing the
default parking lot. The extension is set with the parkext option just as the
default parking lot is done. Also, the manager action has been updated to
optionally allow a specified parking lot.
(closes issue #14882)
Reported by: vmikhnevych
Patches:
patch_14882.txt uploaded by mnick (license 874)
modified by me
Review: https://reviewboard.asterisk.org/r/884/
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Richard Mudgett [Wed, 15 Sep 2010 18:30:54 +0000 (18:30 +0000)]
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r286904 | rmudgett | 2010-09-15 13:28:05 -0500 (Wed, 15 Sep 2010) | 12 lines
Unable to originate calls using E&M over T1.
When originating a call from Unit Under Test to Reference Unit using E&M
RBS signaling mode, I get the following warning message: "Ring/Off-hook in
strange state 3 on channel 1".
Fixed the sig_analog outgoing flag. It was never set when sig_analog was
extracted from chan_dahdi.
JIRA SWP-2191
JIRA AST-408
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r286905 | rmudgett | 2010-09-15 13:29:21 -0500 (Wed, 15 Sep 2010) | 1 line
Simplify some code in sig_analog.
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Matthew Nicholson [Wed, 15 Sep 2010 13:10:50 +0000 (13:10 +0000)]
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r286868 | mnicholson | 2010-09-15 08:05:52 -0500 (Wed, 15 Sep 2010) | 16 lines
Set tohost to the domain specified in the configuration file instead of the IP address of the host we are calling.
This fixes a regression introduced in r274783.
(closes issue #17960)
Reported by: adriavidal
Patches:
sip-tohost-fix1.diff uploaded by mnicholson (license 96)
Tested by: mich, mnicholson, adriavidal
(closes issue #17676)
Reported by: outcast
Patches:
sip-tohost-fix1.diff uploaded by mnicholson (license 96)
Tested by: mnicholson
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David Vossel [Tue, 14 Sep 2010 22:02:00 +0000 (22:02 +0000)]
Merged revisions 286834 via svnmerge from
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r286834 | dvossel | 2010-09-14 16:57:35 -0500 (Tue, 14 Sep 2010) | 2 lines
Sets subscribed type for outgoing MWI subscriptions so correct Event header is used.
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Matthew Nicholson [Tue, 14 Sep 2010 19:29:43 +0000 (19:29 +0000)]
Merged revisions 286758 via svnmerge from
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r286758 | mnicholson | 2010-09-14 14:28:38 -0500 (Tue, 14 Sep 2010) | 27 lines
Merged revisions 286757 via svnmerge from
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r286757 | mnicholson | 2010-09-14 14:27:28 -0500 (Tue, 14 Sep 2010) | 20 lines
Merged revisions 286756 via svnmerge from
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r286756 | mnicholson | 2010-09-14 14:26:18 -0500 (Tue, 14 Sep 2010) | 13 lines
Don't clear the username from a realtime database when a registration expires.
Non-realtime chan_sip does not clear the username from memory when a registration expiries so realtime probably shouldn't either.
(closes issue #17551)
Reported by: ricardolandim
Patches:
reg-expiry-username-1.4-fix1.diff uploaded by mnicholson (license 96)
reg-expiry-username-1.6.2-fix1.diff uploaded by mnicholson (license 96)
reg-expiry-username-1.8-fix1.diff uploaded by mnicholson (license 96)
reg-expiry-username-trunk-fix1.diff uploaded by mnicholson (license 96)
Tested by: ricardolandim, mnicholson
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Matthew Nicholson [Tue, 14 Sep 2010 18:05:39 +0000 (18:05 +0000)]
Merged revisions 286682 via svnmerge from
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r286682 | mnicholson | 2010-09-14 13:04:21 -0500 (Tue, 14 Sep 2010) | 21 lines
Merged revisions 286681 via svnmerge from
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r286681 | mnicholson | 2010-09-14 13:02:24 -0500 (Tue, 14 Sep 2010) | 14 lines
Merged revisions 286679 via svnmerge from
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r286679 | mnicholson | 2010-09-14 13:00:01 -0500 (Tue, 14 Sep 2010) | 7 lines
Only drop duplicate answer frames if the channel is bridged.
Back in r3710 ast_read() was modified to drop answer frames on channels that were in the UP state. This modification prevented bridges that were up before the answer from being broken and reestablished by an ANSWER control frame. That change also prevents pickup of channels called from the ast_dial framework from working properly. The ast_dial framework expects to see an ANSWER frame after dialing and the pickup code queues one but ast_read() drops it. This new change only drops ANSWER frames when the channel is bridged, allowing the answer queued by the pickup code to properly pass through ast_read() on to the ast_dial framework.
ABE-2473
(related to issue #2342)
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Richard Mudgett [Tue, 14 Sep 2010 15:31:51 +0000 (15:31 +0000)]
Merged revisions 286647 via svnmerge from
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r286647 | rmudgett | 2010-09-14 10:30:49 -0500 (Tue, 14 Sep 2010) | 1 line
Corrected documented CONNECTED_LINE and REDIRECTING party manipulation macro names.
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Jan Kalab [Tue, 14 Sep 2010 06:58:43 +0000 (06:58 +0000)]
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r286617 | pitel | 2010-09-14 08:55:44 +0200 (Út, 14 zář 2010) | 7 lines
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 05:08:18 +0000 (05:08 +0000)]
Merged revisions 286588 via svnmerge from
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r286588 | tilghman | 2010-09-14 00:07:16 -0500 (Tue, 14 Sep 2010) | 9 lines
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 23:51:32 +0000 (23:51 +0000)]
Merged revisions 286558 via svnmerge from
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r286558 | tilghman | 2010-09-13 18:50:34 -0500 (Mon, 13 Sep 2010) | 9 lines
Merged revisions 286557 via svnmerge from
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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 23:15:50 +0000 (23:15 +0000)]
Merged revisions 286528 via svnmerge from
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r286528 | tilghman | 2010-09-13 18:12:21 -0500 (Mon, 13 Sep 2010) | 9 lines
Merged revisions 286527 via svnmerge from
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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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Russell Bryant [Mon, 13 Sep 2010 22:13:27 +0000 (22:13 +0000)]
Merged revisions 286112 via svnmerge from
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r286112 | russell | 2010-09-10 15:31:58 -0500 (Fri, 10 Sep 2010) | 9 lines
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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Jason Parker [Mon, 13 Sep 2010 19:40:42 +0000 (19:40 +0000)]
Merged revisions 286457 via svnmerge from
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r286457 | qwell | 2010-09-13 14:40:05 -0500 (Mon, 13 Sep 2010) | 12 lines
Merged revisions 286456 via svnmerge from
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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 15:53:17 +0000 (15:53 +0000)]
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r286426 | rmudgett | 2010-09-13 10:52:14 -0500 (Mon, 13 Sep 2010) | 1 line
Update chan_dahdi.conf.sample to reflect new libpri T309 default value.
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Olle Johansson [Sat, 11 Sep 2010 17:35:15 +0000 (17:35 +0000)]
Whitespace cleanup
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286342
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Olle Johansson [Sat, 11 Sep 2010 17:31:42 +0000 (17:31 +0000)]
Whitespace cleanup and reformatting with { and }
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Olle Johansson [Sat, 11 Sep 2010 17:12:58 +0000 (17:12 +0000)]
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r286270 | oej | 2010-09-11 19:09:22 +0200 (Lör, 11 Sep 2010) | 18 lines
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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Olle Johansson [Sat, 11 Sep 2010 17:10:54 +0000 (17:10 +0000)]
Formatting changes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286271
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Terry Wilson [Fri, 10 Sep 2010 22:15:47 +0000 (22:15 +0000)]
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r286189 | twilson | 2010-09-10 17:04:53 -0500 (Fri, 10 Sep 2010) | 30 lines
Merged revisions 286115 via svnmerge from
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r286115 | twilson | 2010-09-10 15:35:25 -0500 (Fri, 10 Sep 2010) | 23 lines
Merged revisions 286059 via svnmerge from
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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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