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r325545 | mnicholson | 2011-06-29 11:18:39 -0500 (Wed, 29 Jun 2011) | 2 lines
make framehooks prevent native bridging (for real this time)
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r325537 | mnicholson | 2011-06-29 10:34:47 -0500 (Wed, 29 Jun 2011) | 2 lines
don't do native/remote bridging if a framehook is active on the channel
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* Added general option negative_penalty_invalid default off. when set
members are seen as invalid/logged out when there penalty is negative.
for realtime members when set remove from queue will set penalty to -1.
* Added queue option autopausedelay when autopause is enabled it will be
delayed for this number of seconds since last successful call if there
was no prior call the agent will be autopaused immediately.
* Added member option ignorebusy this when set and ringinuse is not
will allow per member control of multiple calls as ringinuse does for
the Queue.
- Mark QUEUE_MEMBER_PENALTY Deprecated it never worked for realtime members
- QUEUE_MEMBER is now R/W supporting setting paused, ignorebusy and penalty.
(closes issue ASTERISK-17421)
(closes issue ASTERISK-17391)
Reported by: irroot
Tested by: irroot, jrose
Review: https://reviewboard.asterisk.org/r/1119/
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r325416 | kpfleming | 2011-06-28 16:50:43 -0500 (Tue, 28 Jun 2011) | 3 lines
Fix random misspelling noticed on asterisk-users.
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r325339 | dvossel | 2011-06-28 15:31:00 -0500 (Tue, 28 Jun 2011) | 4 lines
Fixes locking inversion caused by holding sip pvt lock during async_goto.
(closes ASTERISK-17352)
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r325212 | rmudgett | 2011-06-28 12:30:16 -0500 (Tue, 28 Jun 2011) | 7 lines
Use the device name and not the channel name to initialize the device state.
Correct ASTERISK-11323 implementation as I don't see how it ever worked as
claimed when it used the channel name and not the device name.
(issue ASTERISK-11323)
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r325152 | jrose | 2011-06-28 10:46:29 -0500 (Tue, 28 Jun 2011) | 5 lines
Fixes moh reload breaking custom mode moh classes when the config file is untouched
(closes issue ASTERISK-17730)
Reported by: sdolloff
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If a SDP does not modify the session, we ignore it. However, we were defaulting
no text and video support to true before checking to see if the sdp modified
anything or not. This would result in process_sdp ignoring an sdp but removing
video and text from the call during direct media reinvites.
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r325091 | lmadsen | 2011-06-28 10:12:00 -0500 (Tue, 28 Jun 2011) | 1 line
Remove line from prep_tarball that kills mkrelease.
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r324955 | tilghman | 2011-06-27 11:30:50 -0500 (Mon, 27 Jun 2011) | 5 lines
Save and restore errno from within signal handlers.
This is recommended by the POSIX standard, as well as by the sigaction(2) manpage
for various platforms that we support (e.g. Mac OS X).
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r324914 | rmudgett | 2011-06-27 10:37:19 -0500 (Mon, 27 Jun 2011) | 21 lines
When subscribing MWI to an unsolicited mailbox the first notification is incorrect.
A remote peer subscribed to MWI with the unsolicited option and a local
phone subscribed to the remote mailbox. The notify message-summary events
are sent correctly except for the first one when subscribing, which will
always be 0. This means the phone MWI indicator will be wrong until the
mailbox read/unread count changes and the event is fired.
Looks like this is a regression from ASTERISK-16149.
* Fix the logic to check the cache and if allowed then fallback to
manually counting mailbox messages.
(closes issue ASTERISK-17997)
Reported by: rsw686
Patches:
jira_asterisk_17997_v1.8.patch (license #5621) uploaded by rmudgett
Tested by: rsw686
JIRA SWP-3551
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r324849 | rmudgett | 2011-06-24 15:46:01 -0500 (Fri, 24 Jun 2011) | 15 lines
Syntax errors in dialplan do not display the file name.
When issuing the CLI command "dialplan reload" syntax errors and warnings
are displayed on the console. The offending line number is displayed on
the console, but the file name is not displayed. Errors caught in
main/config.c do display the file name.
(closes issue ASTERISK-17985)
Reported by: ulogic
Patches:
pbx_config.patch uploaded by ulogic (License #5685) modified format
Tested by: rmudgett
JIRA SWP-3554
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r324768 | jrose | 2011-06-24 11:48:06 -0500 (Fri, 24 Jun 2011) | 11 lines
DTMF wasn't being logged on connected consoles when enabled in logger.conf
Previously in order for DTMF to be logged in a connected console session, the user would
have to do logger set channel DTMF on. This corrects that so that it is on by default.
This issue was caused by an off by one error incurred by a logger level count of 6 in
logger.h where it should have been 7.
(closes issue: ASTERISK-17974)
Reported by: Luke H
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There is no reason to clean up features twice.
Review: https://reviewboard.asterisk.org/r/1279/
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r324678 | kmoore | 2011-06-23 13:29:17 -0500 (Thu, 23 Jun 2011) | 11 lines
Merged revisions 324643 via svnmerge from
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r324643 | kmoore | 2011-06-23 13:21:12 -0500 (Thu, 23 Jun 2011) | 4 lines
Addresses AST-2011-008, memory corruption and remote crash in SIP driver.
AST-2011-008
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r324685 | dvossel | 2011-06-23 13:31:00 -0500 (Thu, 23 Jun 2011) | 8 lines
Fixes sip crash when calling remove_uri_parameters with NULL
AST-2011-009
(closes issue ASTERISK-18017)
Reported by: jaredmauch
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r324652 | dvossel | 2011-06-23 13:23:21 -0500 (Thu, 23 Jun 2011) | 20 lines
Merged revisions 324634 via svnmerge from
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r324634 | dvossel | 2011-06-23 13:18:46 -0500 (Thu, 23 Jun 2011) | 13 lines
Merged revisions 324627 via svnmerge from
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r324627 | dvossel | 2011-06-23 13:16:52 -0500 (Thu, 23 Jun 2011) | 7 lines
Addresses AST-2011-010, remote crash in IAX2 driver
Thanks to twilson for identifying the issue and providing the patches.
AST-2011-010
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r324557 | twilson | 2011-06-22 22:10:38 -0500 (Wed, 22 Jun 2011) | 5 lines
Remove tests for parsing address with invalid port
getaddrinfo on OS X returns with EAI_NONAME error when passed a port
greater than 65535. Linux throws no error, so remove the tests for now.
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r324491 | rmudgett | 2011-06-22 14:16:29 -0500 (Wed, 22 Jun 2011) | 1 line
Use correct variable for text SRTP media.
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r324484 | twilson | 2011-06-22 13:52:04 -0500 (Wed, 22 Jun 2011) | 20 lines
Stop sending IPv6 link-local scope-ids in SIP messages
The idea behind the patch listed below was used, but in a more targeted manner.
There are now address stringification functions for addresses that are meant to
be sent to a remote party. Link-local scope-ids only make sense on the machine
from which they originate and so are stripped in the new functions.
There is also a host sanitization function added to chan_sip which is used
for when peer and dialog tohost fields or sip_registry hostnames are used to
craft a SIP message.
Also added are some basic unit tests for netsock2 address parsing.
(closes issue ASTERISK-17711)
Reported by: ch_djalel
Patches:
asterisk-1.8.3.2-ipv6_ll_scope.patch uploaded by ch_djalel (license 1251)
Review: https://reviewboard.asterisk.org/r/1278/
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Also fixed a reference leak in an error path in sip_msg_send().
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r324481 | rmudgett | 2011-06-22 13:41:20 -0500 (Wed, 22 Jun 2011) | 19 lines
Timout or error on INFO or MESSAGE transaction causes call to be lost.
When exchanging INFO messages within a call, 4xx error causes the call to
be disconnected although RFC 2976 explicitly states that such transactions
do not modify the state of the dialog.
When exchanging MESSAGE messages within a call, 4xx error causes the call
to be disconnected. To provide least surprise, we should not disconnect
the call since a MESSAGE is like INFO in this case. (Implied by RFC 3428
Section 2)
(closes issue ASTERISK-17901)
Reported by: neutrino88
Review: https://reviewboard.asterisk.org/r/1257/
Review: https://reviewboard.asterisk.org/r/1258/
JIRA SWP-3486
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r324479 | rmudgett | 2011-06-22 13:26:55 -0500 (Wed, 22 Jun 2011) | 1 line
Comments and whitespace in chan_sip.c
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used in confbridge.
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r324364 | dvossel | 2011-06-21 15:11:52 -0500 (Tue, 21 Jun 2011) | 10 lines
Fixes locking inversion issue in ast_async_goto()
During this function we can not hold the "chan" lock while
doing the masquerade, the explicit goto on the tmp chan, or
the channel alloc. Instead we need to get the channel lock,
store off information about the channel that we need, and
then let the channel lock go for the remainder of the function.
Review: https://reviewboard.asterisk.org/r/1275/
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When playing back a prompt to a channel, confbridge neglects to check for
hangup events causing lockup condititions for hangups that occur before
actually joining the conference. This change ensures that the user is removed
from the conference in the event of a premature hangup.
Review: https://reviewboard.asterisk.org/r/1277/
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r324241 | lmadsen | 2011-06-20 13:12:32 -0500 (Mon, 20 Jun 2011) | 2 lines
Remove extra 'the'.
Reported by Vlad Povorozniuc
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r324237 | twilson | 2011-06-20 12:33:07 -0500 (Mon, 20 Jun 2011) | 12 lines
Ignore media offers with a port of 0
Section 5.1 of RFC3264 states:
A port number of zero in the offer indicates that the stream is offered
but MUST NOT be used.
(closes issue ASTERISK-17845)
Reported by: jacco
Patches:
issue19281_2.patch uploaded by jacco (license 1277)
Tested by: jacco, twilson
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r324178 | lmadsen | 2011-06-17 14:51:16 -0400 (Fri, 17 Jun 2011) | 2 lines
Add Username and Secret fields to manager Login action.
Pointed out by Vlad Povorozniuc
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r324176 | lmadsen | 2011-06-17 14:38:40 -0400 (Fri, 17 Jun 2011) | 2 lines
Fix typo in documentation.
Pointed out by Vlad Povorozniuc
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r324174 | rmudgett | 2011-06-17 13:23:19 -0500 (Fri, 17 Jun 2011) | 5 lines
Add header string to libpri debug output.
Add header string to libpri debug output so the libpri output can be
found/extracted easier from huge debug trace files.
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r324115 | lmadsen | 2011-06-17 11:14:54 -0400 (Fri, 17 Jun 2011) | 3 lines
Fix grammar in documentation for Goto() and GotoIf()
(closes issue ASTERISK-18023)
Reported by: Tim Osman
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r324048 | twilson | 2011-06-16 17:35:41 -0500 (Thu, 16 Jun 2011) | 8 lines
Lock the channel before calling the setoption callback
The channel needs to be locked before calling these callback functions. Also,
sip_setoption needs to lock the pvt and a check p->rtp is non-null before using
it.
Review: https://reviewboard.asterisk.org/r/1220/
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r323990 | rmudgett | 2011-06-16 13:12:32 -0500 (Thu, 16 Jun 2011) | 5 lines
The test_event unit test is occasionally failing.
Wait for the special posted event to process before adding a new
subscription.
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r323932 | twilson | 2011-06-16 10:58:22 -0500 (Thu, 16 Jun 2011) | 4 lines
Don't assume ASTDBDIR exists
It most likely doesn't on FreeBSD
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r323866 | twilson | 2011-06-15 15:03:58 -0500 (Wed, 15 Jun 2011) | 2 lines
Remove now-useless cast of ARRAY_LEN
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r323863 | twilson | 2011-06-15 14:58:18 -0500 (Wed, 15 Jun 2011) | 2 lines
Make ARRAY_LEN() return the same type on x86 and x86_64 systems
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r323859 | twilson | 2011-06-15 14:45:20 -0500 (Wed, 15 Jun 2011) | 2 lines
Fix more ARRAY_LEN format string issues
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r323754 | twilson | 2011-06-15 13:21:52 -0500 (Wed, 15 Jun 2011) | 23 lines
Merged revisions 323733 via svnmerge from
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r323733 | twilson | 2011-06-15 13:13:00 -0500 (Wed, 15 Jun 2011) | 16 lines
Merged revisions 323732 via svnmerge from
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r323732 | twilson | 2011-06-15 13:06:24 -0500 (Wed, 15 Jun 2011) | 9 lines
Fix DYNAMIC_FEATURES
DYNAMIC_FEATURES were broken by a recent DTMF change. This patch makes
sure that dynamic features are also checked when deciding whether or not
to pass DTMF through or store it for interpreting.
(closes issue ASTERISK-17914)
Reported by: vrban
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r323730 | jrose | 2011-06-15 12:42:42 -0500 (Wed, 15 Jun 2011) | 11 lines
Adds locking to find_table in res_configure_pgsql to prevent a crash.
Bryonclark described the problem as occuring during this function because of multiple
simultaneous database operations causing corruption against a pgsqlConn object.
(closes issue ASTERISK-17811)
Reported by: byronclark
Patches:
pgsql_find_table_locking.patch uploaded by byronclark (license 1200)
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r323672 | twilson | 2011-06-15 10:09:51 -0700 (Wed, 15 Jun 2011) | 5 lines
Cast ARRAY_LEN to size_t for ast_logging
32-bit and 64-bit machines return different types for ARRAY_LEN(), so cast
it before using in a format string.
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r323669 | rmudgett | 2011-06-15 11:43:18 -0500 (Wed, 15 Jun 2011) | 21 lines
[regression] Voicemail MWI is no longer sent.
When leaving a voicemail, the MWI message is never sent. The same thing
happens when checking a voicemail and marking it as read.
If you restart Asterisk, everything comes up at that state correctly, but
changes to the messages in voicemail causes the light to not be set
appropriately. Very easy to reproduce.
* Made ast_event_check_subscriber() return TRUE if there are ANY
subscribers to an event type when there are no restricting ie values
passed. This allows an event being queued to be queued.
(closes issue ASTERISK-18002)
Reported by: lmadsen
Tested by: lmadsen, irroot
Patches:
jira_asterisk_18002_v1.8.patch uploaded by rmudgett (License #5621)
(closes issue ASTERISK-18019)
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r323670 | rmudgett | 2011-06-15 11:43:31 -0500 (Wed, 15 Jun 2011) | 7 lines
Add a test to the event unit tests to catch ASTERISK-18002.
The new tests check to see if there are ANY subscribers to the event type
when ast_event_check_subscriber() is not passed any specific ie values.
(issue ASTERISK-18002)
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r323610 | jrose | 2011-06-15 11:09:24 -0500 (Wed, 15 Jun 2011) | 7 lines
Adds PQclear calls on result to various parts of res_conf_pgsql
(closes issue ASTERISK-17812)
Reported by: byronclark
Patches:
pgsql_pqclear.patch uploaded by byronclark (license 1200)
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r323608 | seanbright | 2011-06-15 11:31:53 -0400 (Wed, 15 Jun 2011) | 39 lines
Merged revisions 323579 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r323579 | seanbright | 2011-06-15 11:22:50 -0400 (Wed, 15 Jun 2011) | 32 lines
Merged revisions 323559 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r323559 | seanbright | 2011-06-15 11:15:30 -0400 (Wed, 15 Jun 2011) | 25 lines
Resolve a segfault/bus error when we try to map memory that falls on a page
boundary.
The fix for ASTERISK-15359 was incorrect in that it added 1 to the length of the
mmap'd region. The problem with this is that reading/writing to that extra byte
outside of the bounds of the underlying fd causes a bus error.
The real issue is that we are working with both a FILE * and the raw fd
underneath it and not synchronizing between them. The code that was removed in
ASTERISK-15359 was correct, but we weren't flushing the FILE * before mapping
the fd.
Looking at the manager code in 1.4 reveals that the FILE * in 'struct
mansession' is never used except to create a temporary file that we immediately
fdopen. This means we just need to write a 0 byte to the fd and everything will
just work. The other branches require a call to fflush() which, while not a
guaranteed fix, should reduce the likelihood of a crash.
This all makes sense in my head.
(closes issue ASTERISK-16460)
Reported by: Ravelomanantsoa Hoby (hoby)
Patches:
issue17747_1.4_svn_markII.patch uploaded by Sean Bright (license #5060)
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Added the CONFBRIDGE_INFO dialplan function to get information about a
conference bridge including locked status and number of parties, admins, and
marked users.
Review: https://reviewboard.asterisk.org/r/1271/
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r323456 | rmudgett | 2011-06-14 19:50:20 -0500 (Tue, 14 Jun 2011) | 1 line
Add missing break in ast_event_get_cached().
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r323392 | rmudgett | 2011-06-14 12:21:24 -0500 (Tue, 14 Jun 2011) | 6 lines
Add more strict hostname checking to ast_dnsmgr_lookup().
Change suggested in review.
Review: https://reviewboard.asterisk.org/r/1240/
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r323394 | rmudgett | 2011-06-14 12:21:39 -0500 (Tue, 14 Jun 2011) | 2 lines
Made ast_sockaddr_split_hostport() port warning msgs more meaningful.
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r323370 | twilson | 2011-06-14 09:33:55 -0700 (Tue, 14 Jun 2011) | 10 lines
Add rtpkeepalives back to 1.8
The RTP-engine conversion left out support for handling rtpkeepalives.
This patch adds them back.
(closes issue ASTERISK-17304)
Reported by: lmadsen
Review: https://reviewboard.asterisk.org/r/1226/
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