[asterisk-commits] phsultan: branch phsultan/jabberreceive r176497 - in /team/phsultan/jabberrec...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Feb 17 04:08:27 CST 2009
Author: phsultan
Date: Tue Feb 17 04:08:26 2009
New Revision: 176497
URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=176497
Log:
Merged revisions 171757,171793,171797,171838,171880,171924-171925,171964,172063,172099,172131-172132,172173,172234,172268,172270-172271,172315,172318-172319,172370,172400,172440-172441,172548,172580-172581,172598,172640,172706,172741,172778,172816-172818,172855,172890,172894,172929,172963,173028,173047,173067,173104,173130,173169,173249,173311,173354,173393,173397,173458,173500,173502-173503,173507,173589,173593,173657,173693,173697,173771,173773,173776,173848,173858,173901-173902,173952,173974,174041,174046,174084,174149,174219,174301,174325,174327,174370,174432,174435,174470,174503,174543,174580,174584,174645,174705,174710,174764,174805,174844,174886,174945,174948,174951,175058,175089,175121,175125,175127,175188,175250,175255,175295,175298,175334,175344,175368,175408,175411,175475,175508,175512,175549,175591,175597,175623,175636,175655,175663,175699,175783,175801,175827,175829,175882,175952,175983,176030,176100,176138,176174,176248,176253,176255,176320,176355-176356,176360,176459 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r171757 | dvossel | 2009-01-27 23:43:36 +0100 (Tue, 27 Jan 2009) | 7 lines
Adding AES_ENCRYPT and AES_DECRYPT dialplan functions.
(closes issue #14301)
Reported by: amorsen
review: http://reviewboard.digium.com/r/128/
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r171793 | mattf | 2009-01-28 00:28:51 +0100 (Wed, 28 Jan 2009) | 1 line
Don't complain about lack of D-channels on PTMP connections
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r171797 | mmichelson | 2009-01-28 01:17:55 +0100 (Wed, 28 Jan 2009) | 3 lines
Fix some signedness problems in func_aes.c
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r171838 | oej | 2009-01-28 14:11:44 +0100 (Wed, 28 Jan 2009) | 10 lines
Merged revisions 171837 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r171837 | oej | 2009-01-28 14:07:27 +0100 (Ons, 28 Jan 2009) | 2 lines
Add a better explanation of the difference between the device namespace and the dialplan for newbies.
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r171880 | oej | 2009-01-28 14:26:31 +0100 (Wed, 28 Jan 2009) | 2 lines
Add some more notes about device matching.
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r171924 | oej | 2009-01-28 15:37:16 +0100 (Wed, 28 Jan 2009) | 6 lines
Add final part of previously committed work for answered elsewhere in queue - the missing piece that started with app_dial() earlier on.
This is to avoid having the list and counter of missed calls being touched by queue calls. Add the C option to queue() and nothing
will be logged on phones that support the Reason: header on SIP cancel, like the SNOM phones.
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r171925 | oej | 2009-01-28 15:39:26 +0100 (Wed, 28 Jan 2009) | 2 lines
Yep. Documentation is important.
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r171964 | tilghman | 2009-01-28 18:27:40 +0100 (Wed, 28 Jan 2009) | 9 lines
Merged revisions 171963 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r171963 | tilghman | 2009-01-28 11:25:18 -0600 (Wed, 28 Jan 2009) | 2 lines
Clarify log message (suggested by manxpower on #asterisk-dev)
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r172063 | murf | 2009-01-28 21:31:06 +0100 (Wed, 28 Jan 2009) | 52 lines
Merged revisions 172030 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r172030 | murf | 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) | 46 lines
This patch fixes h-exten running misbehavior in manager-redirected
situations.
What it does:
1. A new Flag value is defined in include/asterisk/channel.h,
AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the
bridge hangup exten code not to run the h-exten there (nor
publish the bridge cdr there). It will done at the pbx-loop
level instead.
2. In the manager Redirect code, I set this flag on the channel
if the channel has a non-null pbx pointer. I did the same for the
second (chan2) channel, which gets run if name2 is set...
and the first succeeds.
3. I restored the ending of the cdr for the pbx loop h-exten
running code. Don't know why it was removed in the first place.
4. The first attempt at the fix for this bug was to place code
directly in the async_goto routine, which was called from a
large number of places, and could affect a large number of
cases, so I tested that fix against a fair number of transfer
scenarios, both with and without the patch. In the process,
I saw that putting the fix in async_goto seemed not to affect
any of the blind or attended scenarios, but still, I was
was highly concerned that some other scenarios I had not tested
might be negatively impacted, so I refined the patch to
its current scope, and jmls tested both. In the process, tho,
I saw that blind xfers in one situation, when the one-touch
blind-xfer feature is used by the peer, we got strange
h-exten behavior. So, I inserted code to swap CDRs and
to set the HANGUP_DONT field, to get uniform behavior.
5. I added code to the bridge to obey the HANGUP_DONT flag,
skipping both publishing the bridge CDR, and running
the h-exten; they will be done at the pbx-loop (higher)
level instead.
6. I removed all the debug logs from the patch before committing.
7. I moved the AUTOLOOP set/reset in the h-exten code in res_features
so it's only done if the h-exten is going to be run. A very
minor performance improvement, but technically correct.
(closes issue #14241)
Reported by: jmls
Patches:
14241_redirect_no_bridgeCDR_or_h_exten_via_transfer uploaded by murf (license 17)
Tested by: murf, jmls
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r172099 | murf | 2009-01-28 22:48:37 +0100 (Wed, 28 Jan 2009) | 1 line
my gcc (Ubuntu 4.3.2-1ubuntu11) 4.3.2 didn't like the \%ld and issued a warning, breaking my dev-mode build. This fixes it.
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r172131 | tilghman | 2009-01-28 23:48:01 +0100 (Wed, 28 Jan 2009) | 7 lines
Fix how we skip fields (to avoid fields which don't exist) when doing an UPDATE.
(closes issue #14205)
Reported by: maxgo
Patches:
20090128__bug14205__5.diff.txt uploaded by Corydon76 (license 14)
Tested by: blitzrage
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r172132 | murf | 2009-01-28 23:52:06 +0100 (Wed, 28 Jan 2009) | 1 line
A further correction: cast the sizeof to an int.
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r172173 | oej | 2009-01-29 10:18:01 +0100 (Thu, 29 Jan 2009) | 24 lines
Merged revisions 172169 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r172169 | oej | 2009-01-29 09:48:18 +0100 (Tor, 29 Jan 2009) | 16 lines
Make sure that we always add the hangupcause headers. In some cases, the owner was disconnected before we checked for the cause.
This patch implements a temporary storage in the pvt and use that instead.
The code is based on ideas from code from Adomjan in issue #13385 (Add support for Reason: header)
Thanks to Klaus Darillion for testing!
(closes issue #14294)
related to issue #13385
Reported by: klaus3000 and adomjan
Patches:
bug14294b.diff uploaded by oej (license 306)
Based on 20080829_chan_sip.c-q850reason_header.patch uploaded by adomjan (license 487)
Tested by: oej, klaus3000
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r172234 | oej | 2009-01-29 12:19:29 +0100 (Thu, 29 Jan 2009) | 7 lines
Make sure register= line supports both port and expiry at the same time.
(closes issue #14185)
Reported by: Nick_Lewis
Patches:
chan_sip.c-expiryrequest6.patch uploaded by Nick (license 657)
Tested by: Nick_Lewis
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r172268 | oej | 2009-01-29 14:21:31 +0100 (Thu, 29 Jan 2009) | 15 lines
- Make sure we set setvar= variables on outbound calls too, not only inbound calls.
- Also, change a function in app.c to return a userful value instead of always returning 0.
Patch by fnordian, changed by Corydon76 and myself.
This does not close the bug report, as fnordian had an additional change we're still discussing.
(related to issue #14059)
Reported by: fnordian
Patches:
chan_sip_hfield.patch uploaded by fnordian (license 110)
20090116__bug14059.diff.txt uploaded by Corydon76 (license 14)
Tested by: fnordian, Corydon76, oej
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r172270 | oej | 2009-01-29 14:24:01 +0100 (Thu, 29 Jan 2009) | 2 lines
Update documentation
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r172271 | lmadsen | 2009-01-29 14:47:27 +0100 (Thu, 29 Jan 2009) | 5 lines
The realtime_pgsql.sql script is missing a couple of fields.
closes issue #14339)
Reported by: fiddur
Patches:
realtime_pgsql.sql.diff uploaded by fiddur (license 678)
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r172315 | tilghman | 2009-01-29 17:48:25 +0100 (Thu, 29 Jan 2009) | 2 lines
Better document mode=multirow, based upon a conversation with Jared.
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r172318 | oej | 2009-01-29 18:08:22 +0100 (Thu, 29 Jan 2009) | 5 lines
Fix "cancel answered elsewhere" through app_queue with members in chan_local.
Also, implement a private cause code (as suggested by Tilghman). This works with
chan_sip, but doesn't propagate through chan_local.
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r172319 | oej | 2009-01-29 18:10:43 +0100 (Thu, 29 Jan 2009) | 2 lines
Revert two lines that was extra, but only on fridays.
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r172370 | rmudgett | 2009-01-29 20:34:09 +0100 (Thu, 29 Jan 2009) | 1 line
Fixed some doxygen comments
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r172400 | rmudgett | 2009-01-29 21:38:34 +0100 (Thu, 29 Jan 2009) | 12 lines
channels/chan_dahdi.c
* Added doxygen comments to the major dahdi structures.
* Fixed PRI and SS7 using an incorrect string value if the extension
delimiter is not present in the Dial() function.
* Fixed SS7 not checking if the dialed extension is at least as long
as the stripmsd option.
* Fixed PRI not handling unknown TON/NPI prefix letters correctly.
* Fixed some uninitialized string variables on FXS ports.
configs/chan_dahdi.conf.sample
* Updated some documentation.
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r172440 | rmudgett | 2009-01-30 00:15:20 +0100 (Fri, 30 Jan 2009) | 1 line
Remove tabs from comment
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r172441 | tilghman | 2009-01-30 00:15:40 +0100 (Fri, 30 Jan 2009) | 16 lines
Merged revisions 172438 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r172438 | tilghman | 2009-01-29 16:54:29 -0600 (Thu, 29 Jan 2009) | 9 lines
Lose the CAP_NET_ADMIN at every fork, instead of at startup. Otherwise, if
Asterisk runs as a non-root user and the administrator does a 'restart now',
Asterisk loses the ability to set QOS on packets.
(closes issue #14004)
Reported by: nemo
Patches:
20090105__bug14004.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76
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r172548 | tilghman | 2009-01-30 19:36:56 +0100 (Fri, 30 Jan 2009) | 2 lines
Parameter position reversed in documentation
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r172580 | twilson | 2009-01-30 22:29:12 +0100 (Fri, 30 Jan 2009) | 44 lines
Merged revisions 172517 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) | 37 lines
Fix feature inheritance with builtin features
When using builtin features like parking and transfers, the AST_FEATURE_* flags
would not be set correctly for all instances when either performing a builtin
attended transfer, or parking a call and getting the timeout callback. Also,
there was no way on a per-call basis to specify what features someone should
have on picking up a parked call (since that doesn't involve the Dial() command).
There was a global option for setting whether or not all users who pickup a
parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT,
AUTOMON, or PARKCALL.
This patch:
1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the
dialplan or with setvar in channels that support it. This variable can be set
to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the
equivalent dial options), to set what features should be activated on this
channel. The patch moves the setting of the features datastores into the
bridging code instead of app_dial to help facilitate this.
2) adds global options parkedcallparking, parkedcallhangup, and
parkedcallrecording to be similar to the parkedcalltransfers option for
globally setting features.
3) has builtin_atxfer call builtin_parkcall if being transfered to the parking
extension since tracking everything through multiple masquerades, etc. is
difficult and error-prone
4) attempts to fix all cases of return calls from parking and completed builtin
transfers not having the correct permissions
(closes issue #14274)
Reported by: aragon
Patches:
fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396)
Tested by: aragon, otherwiseguy
Review http://reviewboard.digium.com/r/138/
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r172581 | twilson | 2009-01-30 22:50:03 +0100 (Fri, 30 Jan 2009) | 2 lines
Remove incorrect line from sample config
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r172598 | mmichelson | 2009-01-30 23:22:04 +0100 (Fri, 30 Jan 2009) | 3 lines
Fix redefinition of flag in channel.h
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r172640 | twilson | 2009-01-31 01:16:46 +0100 (Sat, 31 Jan 2009) | 11 lines
Blocked revisions 172639 via svnmerge
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r172639 | twilson | 2009-01-30 18:15:09 -0600 (Fri, 30 Jan 2009) | 5 lines
Rename new parkedcallparking option to parkedcallreparking
Since this option actually already existed in 1.6.0+, use the same name so as
not to confuse people when they upgrade
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r172706 | tilghman | 2009-01-31 17:40:59 +0100 (Sat, 31 Jan 2009) | 7 lines
Don't increment the loop, now that incrementing is taken care of by the
decoder function.
(closes issue #14363)
Reported by: andrew53
Patches:
func_strings_filter.patch uploaded by andrew53 (license 519)
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r172741 | tilghman | 2009-02-01 03:44:23 +0100 (Sun, 01 Feb 2009) | 4 lines
Blank argument crashes Asterisk
(closes issue #14377)
Reported by: amorsen
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r172778 | seanbright | 2009-02-02 02:41:29 +0100 (Mon, 02 Feb 2009) | 4 lines
The CID lookup feature wasn't actually working properly with dialog-info+xml
supporting devices. The devices (snoms, specifically) need to receive a SIP
URI instead of just an extension. This adds that functionality.
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r172816 | oej | 2009-02-02 11:29:07 +0100 (Mon, 02 Feb 2009) | 3 lines
Add some well-needed improvements to the wishlist in the code, so that we can close
some bug reports.
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r172817 | oej | 2009-02-02 11:44:48 +0100 (Mon, 02 Feb 2009) | 2 lines
Small formatting change
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r172818 | oej | 2009-02-02 11:46:19 +0100 (Mon, 02 Feb 2009) | 3 lines
Add a todo. I do need to really check what's going on with this kill-the-user business ;-)
Why do we suddenly have two flags to set peer type?
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r172855 | russell | 2009-02-02 17:42:58 +0100 (Mon, 02 Feb 2009) | 2 lines
Fix a spelling mistake.
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r172890 | murf | 2009-02-02 18:37:15 +0100 (Mon, 02 Feb 2009) | 41 lines
This change allows the disconnect feature (as in "one-touch" in features.c)
to be used within the dial app, before a call is bridged.
Many thanks to sobomax for submitting this patch.
Quoting from bug 11582:
"So the goal of the patch was to use the user configured feature code during the
call setup phase. The original ast_feature_interpret() function is not well suited
for this purpose as it uses much call bridge specific data and doesn't separate a
detection of feature from a feature handler call. So a new function ast_feature_detect()
has been extracted off the ast_feature_interpret() function but keeping the original
logic intact except some insignificant changes to locking.
"Having created the ast_feature_detect() function the possibility to use feature detection
in almost any place of the asterisk code. So a call to this function has been added to
wait_for_answer() function of app_dial.so module. This code doesn't call the feature handler
however and uses old call leg disconnect logic to make the changes as small and simple as
possible to prevent unexpected problems. A disconnect feature currently is the only one
supported during call setup as other features as call parking and call transfer don't make much
sense during call setup. However if need in some of the features would arise it is much easier to
implement as the infrastructure changes are already in place with this patch."
I have cleaned up the patch somewhat, and verified that the existing functionality is not
harmed, and that the new functionality works. Terry has committed his stuff, and there were
no conflicts (see 14274).
(closes issue #11583)
Reported by: sobomax
Patches:
patch-apps__app_dial.c uploaded by sobomax (license 359)
patch-include__asterisk__features.h uploaded by sobomax (license 359)
patch-res__res_features.c uploaded by sobomax (license 359)
enable-features-during-call-setup.diff uploaded by sobomax (license 359)
11583.newdiff uploaded by murf (license 17)
enable-features-during-call-setup-1.diff uploaded by sobomax (license 359)
11583.latest-patch uploaded by murf (license 17)
Tested by: sobomax, murf
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r172894 | lmadsen | 2009-02-02 19:13:40 +0100 (Mon, 02 Feb 2009) | 7 lines
Update the res_ldap.conf file with a better working example.
(closes issue #13861)
Reported by: scramatte
Patches:
__20080110-res_ldap.conf-2.patch uploaded by blitzrage (license 10)
Tested by: jcovert
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r172929 | murf | 2009-02-02 20:02:24 +0100 (Mon, 02 Feb 2009) | 7 lines
This reverts the changes I made for 11583; will
reviewboard this before committing again...
reopened 11583 until all Russell's issues are
resolved.
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r172963 | rmudgett | 2009-02-02 21:40:27 +0100 (Mon, 02 Feb 2009) | 18 lines
Recorded merge of revisions 172962 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r172962 | rmudgett | 2009-02-02 14:28:54 -0600 (Mon, 02 Feb 2009) | 11 lines
channels/chan_dahdi.c
* Added doxygen comments to the major dahdi structures.
* Fixed PRI using an incorrect string value if the extension
delimiter is not present in the Dial() function.
* Fixed some uninitialized string variables on FXS ports.
configs/chan_dahdi.conf.sample
* Updated some documentation.
These changes are already in trunk -r172400
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r173028 | mmichelson | 2009-02-03 00:10:47 +0100 (Tue, 03 Feb 2009) | 9 lines
Add a CLI command to log out a manager user
(closes issue #13877)
Reported by: eliel
Patches:
cli_manager_logout.patch.txt uploaded by eliel (license 64)
Tested by: eliel, putnopvut
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r173047 | mmichelson | 2009-02-03 00:21:33 +0100 (Tue, 03 Feb 2009) | 4 lines
Reverting commit number 173028 as there are some
potential issues
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r173067 | twilson | 2009-02-03 00:57:25 +0100 (Tue, 03 Feb 2009) | 9 lines
Merged revisions 173066 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r173066 | twilson | 2009-02-02 17:48:06 -0600 (Mon, 02 Feb 2009) | 2 lines
Fix a feature inheritance bug I added after code review
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r173104 | tilghman | 2009-02-03 01:24:52 +0100 (Tue, 03 Feb 2009) | 12 lines
Merged revisions 173070 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r173070 | tilghman | 2009-02-02 18:15:59 -0600 (Mon, 02 Feb 2009) | 5 lines
Add warning to standard config, that globals may be overridden by other
dialplan configuration files.
(closes issue #14388)
Reported by: macli
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r173130 | tilghman | 2009-02-03 01:29:49 +0100 (Tue, 03 Feb 2009) | 7 lines
1. Make OS X compile cleanly with app_stack.
2. Use curl to download sound files, as curl is installed natively on OS X,
whereas wget and fetch are not.
(closes issue #14332)
Reported by: oej
Tested by: Corydon76
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r173169 | rmudgett | 2009-02-03 18:35:37 +0100 (Tue, 03 Feb 2009) | 2 lines
Broke up the large conditional blocks so it is easy to see if a function is compiled.
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r173249 | dvossel | 2009-02-04 00:39:14 +0100 (Wed, 04 Feb 2009) | 13 lines
Blocked revisions 173248 via svnmerge
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r173248 | dvossel | 2009-02-03 17:35:55 -0600 (Tue, 03 Feb 2009) | 6 lines
Fixes issue with IAX2 transfer not handing off calls.
Fixes issue with IAX2 transfers not taking place. As it was, a call that was being transfered would never be handed off correctly to the call ends because of how call numbers were stored in a hash table. The hash table, "iax_peercallno_pvt", storing all the current call numbers did not take into account the complications associated with transferring a call, so a separate hash table was required. This second hash table "iax_transfercallno_pvt" handles calls being transfered, once the call transfer is complete the call is removed from the transfer hash table and added to the peer hash table resuming normal operations. Addition functions were created to handle storing, removing, and comparing items in the iax_transfercallno_pvt table.
(issue #13468)
Review: http://reviewboard.digium.com/r/140/
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r173311 | tilghman | 2009-02-04 01:43:52 +0100 (Wed, 04 Feb 2009) | 10 lines
Ensure that commas placed in the middle of extension character classes do not
interfere with correct parsing of the extension. Also, if an unterminated
character class DOES make its way into the pbx core (through some other
method), ensure that it does not crash Asterisk.
(closes issue #14362)
Reported by: Nick_Lewis
Patches:
20090129__bug14362.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76
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r173354 | mmichelson | 2009-02-04 16:30:12 +0100 (Wed, 04 Feb 2009) | 30 lines
Fix a problem where file playback would cause fds to remain open forever
The problem came from the fact that a frame read from a format interpreter
was not freed. Adding a call to ast_frfree fixed this. The explanation for
why this caused the problem is a bit complex, but here goes:
There was a problem in all versions of Asterisk where the embedded frame
of a filestream structure was referenced after the filestream was freed. This
was fixed by adding reference counting to the filestream structure. The refcount
would increase every time that a filestream's frame pointer was pointing to an
actual frame of data. When the frame was freed, the refcount would decrease. Once
the refcount reached 0, the filestream was freed, and as part of the operation,
the open files were closed as well.
Thus it becomes more clear why a missing ast_frfree would cause a reference leak
and cause the files to not be closed. You may ask then if there was a frame leak
before this patch. The answer to that is actually no! The filestream code was
"smart" enough to know that since the frame we received came from a format interpreter,
the frame had no malloced data and thus didn't need to be freed. Now, however, there
is cleanup that needs to be done when we finish with the frame, so we do need to
call ast_frfree on the frame to be sure that the refcount for the filestream is
decremented appropriately.
(closes issue #14384)
Reported by: fiddur
Patches:
14384.patch uploaded by putnopvut (license 60)
Tested by: fiddur, putnopvut
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r173393 | mmichelson | 2009-02-04 18:41:02 +0100 (Wed, 04 Feb 2009) | 11 lines
Merged revisions 173392 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r173392 | mmichelson | 2009-02-04 11:40:29 -0600 (Wed, 04 Feb 2009) | 3 lines
Add a missing unlock. Extremely unlikely to ever matter, but it's needed.
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r173397 | mmichelson | 2009-02-04 18:45:14 +0100 (Wed, 04 Feb 2009) | 11 lines
Merged revisions 173396 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r173396 | mmichelson | 2009-02-04 11:44:48 -0600 (Wed, 04 Feb 2009) | 3 lines
Revert my previous change because it was stupid
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r173458 | tilghman | 2009-02-04 19:48:06 +0100 (Wed, 04 Feb 2009) | 9 lines
When using a socket as a FILE *, the stdio functions will sometimes try to do
an fseek() on the stream, which is an invalid operation for a socket. Turning
off buffering explicitly lets the stdio functions know they cannot do this,
thus avoiding a potential error.
(closes issue #14400)
Reported by: fnordian
Patches:
tcptls.patch uploaded by fnordian (license 110)
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r173500 | jpeeler | 2009-02-04 22:17:53 +0100 (Wed, 04 Feb 2009) | 23 lines
Merged revisions 173211 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r173211 | jpeeler | 2009-02-03 15:57:01 -0600 (Tue, 03 Feb 2009) | 17 lines
Parking attempts made to one end of a bridge no longer will hang up due to a
parking failure.
Parking attempts made using either one-touch, or doing either a blind or
assisted transfer to the parking extension now keep up the bridge instead of
hanging up the attempted parked party. Normal causes for the parking attempt
to fail includes the specific specified extension (via PARKINGEXTEN) not being
available or if all the parking spaces are currently in use. To avoid having
to reverse a masquerade park_space_reserve was made to provide foresight if
a parking attempt will succeed and if so reserve the parking space.
(closes issue #13494)
Reported by: mdu113
Reviewed by Russell: http://reviewboard.digium.com/r/133/
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r173502 | dvossel | 2009-02-04 22:25:14 +0100 (Wed, 04 Feb 2009) | 9 lines
Fixes issue with IAX2 transfer not handing off calls. Reverts changes in 116884
Fixes issue with IAX2 transfers not taking place. As it was, a call that was being transfered would never be handed off correctly to the call ends because of how call numbers were stored in a hash table. The hash table, "iax_peercallno_pvt", storing all the current call numbers did not take into account the complications associated with transferring a call, so a separate hash table was required. This second hash table "iax_transfercallno_pvt" handles calls being transfered, once the call transfer is complete the call is removed from the transfer hash table and added to the peer hash table resuming normal operations. Addition functions were created to handle storing, removing, and comparing items in the iax_transfercallno_pvt table. The changes reverted in 116884 caused backwards compatibility issues involving iax2 transfer with 1.6.0, 1.4, and 1.2.
(closes issue #13468)
Reported by: nicox
Tested by: dvossel
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r173503 | tilghman | 2009-02-04 22:26:15 +0100 (Wed, 04 Feb 2009) | 6 lines
Add XML documentation for the applications and functions in res_jabber
(closes issue #14405)
Reported by: snuffy
Patches:
xml_jabber.diff uploaded by snuffy (license 35)
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r173507 | mmichelson | 2009-02-04 23:16:19 +0100 (Wed, 04 Feb 2009) | 7 lines
Fix some areas where the incorrect interface was passed to ast_device_state
I swear it feels like I already did this once...
(closes issue #14359)
Reported by: francesco_r
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r173589 | mmichelson | 2009-02-05 19:34:06 +0100 (Thu, 05 Feb 2009) | 33 lines
Merged revisions 173559 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r173559 | mmichelson | 2009-02-05 11:34:33 -0600 (Thu, 05 Feb 2009) | 25 lines
Fix a problem where a channel pointer becomes invalid due to masquerading or hanging up.
app_mixmonitor runs its own thread to monitor the channel's activity and write the mixed
audio to a file. Since this thread runs independently of the channel, it is possible that
the mixmonitor thread's channel pointer will point to freed memory when the channel either
is masqueraded or hangs up (technically, both cases are hangups, but we need to handle the
cases slightly differently).
The solution for this is to employ a datastore, which has the nice benefit of allowing us
to hook into channel masquerades and hangups and update our pointer as necessary. If this
looks familiar, this same technique is employed in app_chanspy. app_chanspy is a bit more
involved since it does a lot more operations on the channel that is being spied upon.
app_mixmonitor does have an extra touch that app_chanspy doesn't have, though. Since there
is a thread race between the channel's thread and the mixmonitor thread on a hangup, we em-
ploy a condition-and-boolean combination to ensure that the channel thread finishes with
our structure before the mixmonitor thread attempts to free it. No crashes!
(closes issue #14374)
Reported by: aragon
Patches:
14374.patch uploaded by putnopvut (license 60)
Tested by: aragon, putnopvut
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r173593 | mmichelson | 2009-02-05 19:48:55 +0100 (Thu, 05 Feb 2009) | 11 lines
Merged revisions 173592 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r173592 | mmichelson | 2009-02-05 12:47:24 -0600 (Thu, 05 Feb 2009) | 3 lines
Add some missing cleanup to app_mixmonitor
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r173657 | tilghman | 2009-02-05 20:36:29 +0100 (Thu, 05 Feb 2009) | 2 lines
Change the first field, or we don't get the necessary field separation.
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r173693 | mmichelson | 2009-02-05 21:30:45 +0100 (Thu, 05 Feb 2009) | 20 lines
Merged revisions 173692 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r173692 | mmichelson | 2009-02-05 14:29:09 -0600 (Thu, 05 Feb 2009) | 12 lines
Fix situations where queue members could be autopaused unexpectedly
Specifically, this patch prevents us from autopausing members when
we receive a busy or congestion frame from them.
(closes issue #14376)
Reported by: fiddur
Patches:
14376.patch uploaded by putnopvut (license 60)
Tested by: fiddur
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r173697 | jpeeler | 2009-02-05 22:00:26 +0100 (Thu, 05 Feb 2009) | 18 lines
Merged revisions 173696 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r173696 | jpeeler | 2009-02-05 14:47:51 -0600 (Thu, 05 Feb 2009) | 12 lines
Add new configuration option to make shared IMAP mailboxes function as expected.
The new option is "imapvmshareid" which is an ID to tag multiple mailboxes
using the same IMAP storage location to function as one mailbox. This allows
all messages to be retrieved for any user in the group. The patch alters the
'X-Asterisk-VM-Extension' header that is responsible for matching voicemails
for a given user.
(closes issue #13673)
Reported by: howardwilkinson
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r173771 | mmichelson | 2009-02-06 00:19:51 +0100 (Fri, 06 Feb 2009) | 27 lines
Blocked revisions 173770 via svnmerge
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r173770 | mmichelson | 2009-02-05 17:19:16 -0600 (Thu, 05 Feb 2009) | 20 lines
Fix logic regarding when to perform an SRV lookup for outgoing REGISTER requests
With this fix, we only will perform an SRV lookup at the following times:
* The first time we register with a remote registrar
* If we send a REGISTER but do not receive a response
* If the sendto() function returns an error
While I wrote the patch that fixes this issue, a huge amount of credit is due
to Brett Bryant, who wrote the initial patch on which I based this one.
(closes issue #12312)
Reported by: jrast
Patches:
12312.patch uploaded by putnopvut (license 60)
Tested by: blitzrage
Review: http://reviewboard.digium.com/r/132/
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r173773 | mmichelson | 2009-02-06 00:28:19 +0100 (Fri, 06 Feb 2009) | 7 lines
Properly set "seen" and "unseen" flags when moving messages from the new to the old folder when using IMAP for voicemail storage
(closes issue #13905)
Reported by: jaroth
Patches:
foldermove_v2.patch uploaded by jaroth (license 50)
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r173776 | mmichelson | 2009-02-06 00:48:48 +0100 (Fri, 06 Feb 2009) | 14 lines
Update extensions.conf.sample to be correct.
In trunk, the only necessary change pointed out was that the call
to ChanIsAvail uses an option that has been removed.
For the 1.6.1 branch, however, it appears that the sample file is
badly in need of updating since there are |'s used all over the place
there. My tentative plan is just to copy trunk's sample config file
to those branches since the info there is most up-to-date and should
be correct for use in 1.6.1
Thanks to macli in #asterisk-dev for bringing this up
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r173848 | russell | 2009-02-06 11:25:09 +0100 (Fri, 06 Feb 2009) | 2 lines
Resolve a memory leak that would occur on an invalid channel given to Action: Status
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r173858 | russell | 2009-02-06 11:55:35 +0100 (Fri, 06 Feb 2009) | 13 lines
Add a common implementation of a scheduler context with a dedicated thread.
This commit expands the Asterisk scheduler API to include a common implementation
of a scheduler context being processed by a dedicated thread. chan_iax2 has been
updated to use this new code. Also, as a result, this resolves some race
conditions related to the previous chan_iax2 scheduler handling.
Related to rev 171452 which resolved the same issues in 1.4.
Code from team/russell/sched_thread2
Review: http://reviewboard.digium.com/r/129/
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r173901 | tilghman | 2009-02-06 16:44:23 +0100 (Fri, 06 Feb 2009) | 9 lines
Blocked revisions 173900 via svnmerge
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r173900 | tilghman | 2009-02-06 09:43:32 -0600 (Fri, 06 Feb 2009) | 3 lines
Backport OS X fix from trunk
(AGAIN, closes issue #14360)
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r173902 | file | 2009-02-06 16:59:17 +0100 (Fri, 06 Feb 2009) | 4 lines
Always detach and destroy the whisper and barge audiohooks. Additionally also allow an audiohook to be detached if it has not been attached.
(closes issue #14414)
Reported by: bluecrow76
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r173952 | mnicholson | 2009-02-06 17:28:19 +0100 (Fri, 06 Feb 2009) | 14 lines
Merged revisions 173917 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r173917 | mnicholson | 2009-02-06 10:20:23 -0600 (Fri, 06 Feb 2009) | 7 lines
Limit the addition of the Contact header in SIP responses according to various
SIP RFCs.
(closes issue #13602)
Reported by: hjourdain
Tested by: mnicholson
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r173974 | file | 2009-02-06 18:18:35 +0100 (Fri, 06 Feb 2009) | 15 lines
Merged revisions 173967-173968 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r173967 | file | 2009-02-06 13:14:15 -0400 (Fri, 06 Feb 2009) | 4 lines
Some clients do not put the call-id for replaces at the beginning, so support it being anywhere in the string.
(closes issue #14350)
Reported by: fhackenberger
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r173968 | file | 2009-02-06 13:15:01 -0400 (Fri, 06 Feb 2009) | 2 lines
Remove a debug message I put in by accident.
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r174041 | file | 2009-02-06 20:28:53 +0100 (Fri, 06 Feb 2009) | 4 lines
Don't subscribe to a mailbox on pseudo channels. It is futile. This solves an issue where duplicated pseudo channels would cause a crash because the first one would unsubscribe and the next one would also try to unsubscribe the same subscription.
(closes issue #14322)
Reported by: amessina
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r174046 | dvossel | 2009-02-06 21:12:33 +0100 (Fri, 06 Feb 2009) | 12 lines
Adds immediate yes/no option to iax.conf
This is very similar to the DAHDI immediate=yes option. When the phone is picked up, instead of giving a dialtone it connects directly to the "s" extension. Changes where implemented in chan_iax2.c to directly connect to the "s" extension in the appropriate context when this option is enabled. Examples explaining its use are added to iax2.conf.sample. CHANGES has been updated as well.
(closes issue #14266)
Reported by: jcovert
Patches:
chan_iax2.c.patch-trunk uploaded by jcovert (license 551)
iax.conf.sample.patch uploaded by jcovert (license 551)
Tested by: jcovert, dvossel
Review: http://reviewboard.digium.com/r/143/
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r174084 | dhubbard | 2009-02-07 00:51:56 +0100 (Sat, 07 Feb 2009) | 22 lines
Merged revisions 174082 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r174082 | dhubbard | 2009-02-06 17:36:03 -0600 (Fri, 06 Feb 2009) | 5 lines
check ast_strlen_zero() before calling ast_strdupa() in sip_uri_headers_cmp()
and sip_uri_params_cmp()
The reporter didn't actually upload a properly-formed patch, instead a
modified chan_sip.c file was uploaded. I created a patch to determine the
changes, then modified the suggested changes to create a proper fix. The
summary above is a complete description of the changes.
(closes issue #13547)
Reported by: tecnoxarxa
Patches:
chan_sip.c.gz uploaded by tecnoxarxa (license 258)
Tested by: tecnoxarxa
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r174149 | russell | 2009-02-07 17:16:50 +0100 (Sat, 07 Feb 2009) | 10 lines
Merged revisions 174148 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r174148 | russell | 2009-02-07 10:15:07 -0600 (Sat, 07 Feb 2009) | 2 lines
Fix a race condition that could cause a crash.
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r174219 | file | 2009-02-09 15:49:24 +0100 (Mon, 09 Feb 2009) | 11 lines
Merged revisions 174218 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r174218 | file | 2009-02-09 10:48:21 -0400 (Mon, 09 Feb 2009) | 4 lines
Don't overwrite our pointer to the music class when music on hold stops. We will use this if it starts again to see if we can resume the music where it left off.
(closes issue #14407)
Reported by: mostyn
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r174301 | mmichelson | 2009-02-09 18:20:55 +0100 (Mon, 09 Feb 2009) | 20 lines
Merged revisions 174282 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r174282 | mmichelson | 2009-02-09 11:11:05 -0600 (Mon, 09 Feb 2009) | 12 lines
Don't do an SRV lookup if a port is specified
RFC 3263 says to do A record lookups on a hostname
if a port has been specified, so that's what we're
going to do. See section 4.2.
(closes issue #14419)
Reported by: klaus3000
Patches:
patch_chan_sip_nosrvifport_1.4.23.txt uploaded by klaus3000 (license 65)
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r174325 | dvossel | 2009-02-09 18:26:02 +0100 (Mon, 09 Feb 2009) | 9 lines
Fixes issue with hangups not being sent and external process never terminating.
The ignore_hangup, run_dead, and noanswer flags were never initilized to zero causing hangups to never be issued. If the external script expects to be notified of a hangup and never receives one, it runs indefinitely.
(closes issue #14251)
Reported by: chris-mac
Tested by: dvossel
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r174327 | mmichelson | 2009-02-09 18:27:32 +0100 (Mon, 09 Feb 2009) | 3 lines
Fix something I messed up in the merge I just did
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r174370 | murf | 2009-02-10 03:45:56 +0100 (Tue, 10 Feb 2009) | 10 lines
Merged revisions 174369 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r174369 | murf | 2009-02-09 19:27:40 -0700 (Mon, 09 Feb 2009) | 5 lines
This patch solves some compiler complaints
in both 32 and 64-bit environments.
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r174432 | murf | 2009-02-10 05:36:22 +0100 (Tue, 10 Feb 2009) | 3 lines
More intptr_t work.
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r174435 | murf | 2009-02-10 05:49:02 +0100 (Tue, 10 Feb 2009) | 8 lines
This patch removes the use of AST_PBX_KEEPALIVE
from app_rpt.c.
(closes issue #14435)
Reported by: D_McNaul
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