https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r313517 | rmudgett | 2011-04-12 17:35:53 -0500 (Tue, 12 Apr 2011) | 12 lines
Bring the dumpchan application inline with "core show channel".
* Added fields that are in "core show channel" to dumpchan output.
* Fixed reuse of formatbuf before the previous string stored there was
used by snprintf. All output strings now have their own buffer.
* Adjusted the buffer sizes to not be so abusive of the stack now that
there are more buffers.
Change requested by oej.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313527 65c4cc65-6c06-0410-ace0-fbb531ad65f3
IPv6 support for ooh323,
bindaddr, peers and users ip can be IPv4 or IPv6 addr
correction for multi-homed mode (0.0.0.0 or :: bindaddr)
can work in dual 6/4 mode with :: bindaddr
gatekeeper mode isn't supported in v6 mode while
(issue #18278)
Reported by: may213
Patches:
ipv6-ooh323.patch uploaded by may213 (license 454)
Review: https://reviewboard.asterisk.org/r/1004/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313482 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
also went ahead and fixed the problem it introduces before committing.
........
r313435 | jrose | 2011-04-12 13:44:44 -0500 (Tue, 12 Apr 2011) | 1 line
fixing stupid mistake with putting code before variable declaration
........
Merged revisions 313433 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r313432 | jrose | 2011-04-12 13:12:29 -0500 (Tue, 12 Apr 2011) | 14 lines
reload Chan_dahdi memory leak caused by variables
chan_dahdi reloading with variables set via setvar in chan_dahdi.conf would
stay in the dahdi_pvt structs for individual channels (causing them to just
continue adding the new ones to the list) and also there was a memory leak
causes by the conf objects. This patch resolves both of these by using
ast_variables_destroy during the loading process.
(closes issue #17450)
Reported by: nahuelgreco
Patches:
patch.diff uploaded by jrose (license 1225)
Tested by: tilghman, jrose
Review: https://reviewboard.asterisk.org/r/1170/
........
........
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313437 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r313368 | rmudgett | 2011-04-11 18:03:02 -0500 (Mon, 11 Apr 2011) | 2 lines
Backport a restructuring change from trunk to make the next change stand out.
........
r313369 | rmudgett | 2011-04-11 18:08:02 -0500 (Mon, 11 Apr 2011) | 13 lines
Frames from the inbound channel should go to all outbound channels in app_dial.c.
In app_dial.c:wait_for_answer() frames from the inbound channel should be
sent to all outbound channels instead of only if there is just one
outbound channel.
Control frames like AST_CONTROL_CONNECTED_LINE need to be passed to all of
the the outbound channels. This can happen if a blond transfer is done by
a remote switch on the inbound channel.
JIRA AST-443
JIRA SWP-2730
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313383 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r313190 | rmudgett | 2011-04-11 10:40:30 -0500 (Mon, 11 Apr 2011) | 39 lines
Merged revisions 313189 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r313189 | rmudgett | 2011-04-11 10:32:53 -0500 (Mon, 11 Apr 2011) | 32 lines
Merged revisions 313188 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r313188 | rmudgett | 2011-04-11 10:27:52 -0500 (Mon, 11 Apr 2011) | 25 lines
Stuck channel using FEATD_MF if caller hangs up at the right time.
The cause was actually a caller hanging up just at the end of the Feature
Group D DTMF tones that setup the call. The reason for this is a "guard
timer" that's implemented using ast_safe_sleep(100). If the caller
happens to hang up AFTER the final tone of the DTMF string but BEFORE the
end of that ast_safe_sleep(), then ast_safe_sleep() will return non-zero.
This causes the code to bounce to the end of ss_thread(), but it does NOT
tear down the call properly.
This should be a rare occurrence because the caller has to hang up at
EXACTLY the right time. Nonetheless, it was happening quite regularly on
the reporter's system. It's not easily reproducible, unless you purposely
increase the guard-time to 2000 or more. Once you do that, you can
reproduce it every time by watching the DTMF debug and hanging up just as
it ends.
Simply add an ast_hangup() before goto quit.
(closes issue #15671)
Reported by: jcromes
Patches:
issue15671.patch uploaded by pabelanger (license 224)
Tested by: jcromes
........
................
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313191 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r313001 | alecdavis | 2011-04-07 22:19:31 +1200 (Thu, 07 Apr 2011) | 13 lines
Fix ISDN calling subaddr User Specified Odd/Even Flag
Calculation of the Odd/Even flag was wrong.
Implement correct algo, and set odd/even=0 if data would be truncated.
Only allow automatic calculation of the O/E flag, don't let dialplan influence.
(closes issue #19062)
Reported by: festr
Patches:
bug19062.diff2.txt uploaded by alecdavis (license 585)
Tested by: festr, alecdavis, rmudgett
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313005 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r312949 | rmudgett | 2011-04-05 13:45:24 -0500 (Tue, 05 Apr 2011) | 6 lines
Crash if ISDN span layer 1 is down on initial load.
Regression from -r312575 B channel shifting during negotiation.
* Also combine updating the alarm flag with clearing the resetting flag.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312950 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r312866 | rmudgett | 2011-04-05 10:38:14 -0500 (Tue, 05 Apr 2011) | 15 lines
Responding to OPTIONS packet with 404 because Asterisk not looking for "s" extension.
The get_destination() function was not using the "s" extension when the
request URI did not specify an extension. This is a regression caused
when the URI parsing code was extracted into parse_uri().
Made get_destination() substitute the "s" extension when the parsed URI
results in an empty string.
(closes issue #18348)
Reported by: shmaize
Patches:
issue18348_v1.8.patch uploaded by rmudgett (license 664)
Tested by: shmaize
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312868 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If the user invokes 'dialplan add extension' into a non-existing context, the context will be created
and a message informing the user of the context being created will be issued in cli.
(closes issue #17431)
Reported by: leearcher
Patches:
context_auto_create.diff uploaded by kobaz (license 834)
Tested by: leearcher, kobaz, jrose
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312678 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r312575 | rmudgett | 2011-04-04 11:10:50 -0500 (Mon, 04 Apr 2011) | 52 lines
Merged revisions 312574 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r312574 | rmudgett | 2011-04-04 11:00:02 -0500 (Mon, 04 Apr 2011) | 45 lines
Merged revisions 312573 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r312573 | rmudgett | 2011-04-04 10:49:30 -0500 (Mon, 04 Apr 2011) | 38 lines
Issues with ISDN calls changing B channels during call negotiations.
The handling of the PROCEEDING message was not using the correct call
structure if the B channel was changed. (The same for PROGRESS.) The call
was also not hungup if the new B channel is not provisioned or is busy.
* Made all call connection messages (SETUP_ACKNOWLEDGE, PROCEEDING,
PROGRESS, ALERTING, CONNECT, CONNECT_ACKNOWLEDGE) ensure that they are
using the correct structure and B channel. If there is any problem with
the operations then the call is now hungup with an appropriate cause code.
* Made miscellaneous messages (INFORMATION, FACILITY, NOTIFY) find the
correct structure by looking for the call and not using the channel ID.
NOTIFY is an exception with versions of libpri before v1.4.11 because a
call pointer is not available for Asterisk to use.
* Made all hangup messages (DISCONNECT, RELEASE, RELEASE_COMPLETE) find
the correct structure by looking for the call and not using the channel
ID.
(closes issue #18313)
Reported by: destiny6628
Tested by: rmudgett
JIRA SWP-2620
(closes issue #18231)
Reported by: destiny6628
Tested by: rmudgett
JIRA SWP-2924
(closes issue #18488)
Reported by: jpokorny
JIRA SWP-2929
JIRA AST-437 (The issues fixed here are most likely causing this JIRA issue.)
JIRA DAHDI-406
JIRA LIBPRI-33 (Stuck resetting flag likely fixed)
........
................
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r312509 | rmudgett | 2011-04-01 18:15:42 -0500 (Fri, 01 Apr 2011) | 22 lines
When a call going out an NT-PTMP port gets rejected, Asterisk crashes.
If a call is sent to an ISDN phone that rejects the call with
RELEASE_COMPLETE(cause: call reject(21), or busy(17)) Asterisk crashes.
I could not get my setup to crash. However, I could see the possibility
from a race condition between queuing an AST_CONTROL_BUSY to the core and
then queueing an AST_CONTROL_HANGUP. If the AST_CONTROL_BUSY is processed
before the AST_CONTROL_HANGUP is queued, the ast_channel could be
destroyed out from under chan_misdn.
Avoid this particular crash scenario by not queueing the
AST_CONTROL_HANGUP if the AST_CONTROL_BUSY was queued.
(closes issue #18408)
Reported by: wimpy
Patches:
issue18408_v1.8.patch uploaded by rmudgett (license 664)
Tested by: rmudgett, wimpy
JIRA SWP-2679
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312510 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r312461 | rmudgett | 2011-04-01 16:31:39 -0500 (Fri, 01 Apr 2011) | 25 lines
CallCompletionRequest()/CallCompletionCancel() exit non-zero if fail.
The CallCompletionRequest()/CallCompletionCancel() dialplan applications
exit nonzero on normal failure conditions. The nonzero exit causes the
dialplan to hangup immediately. The dialplan author has no opportunity to
report success/failure to the user.
* Made always return zero so the dialplan can continue.
* Made set CC_REQUEST_RESULT/CC_REQUEST_REASON and
CC_CANCEL_RESULT/CC_CANCEL_REASON channel variables respectively. Also
documented the values set.
* Reduced the warning about no core instance in CallCompletionCancel() to
a debug message. It is a normal event and should not be output at the
WARNING level.
(closes issue #18763)
Reported by: p_lindheimer
Patches:
ccss.patch uploaded by p lindheimer (license 558) Modified
Tested by: p_lindheimer, rmudgett
JIRA SWP-3042
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312462 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In chan_dahdi.conf, the user can now use length 4 patterns in addition to the usual length 2 patterns. The s
ntax remains the same and the method used to track the pattern history will only change when using the length
4 patterns.
(closes issue SWP-3250)
Code:
jrose
rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r312211 | alecdavis | 2011-04-01 22:03:11 +1300 (Fri, 01 Apr 2011) | 36 lines
Merged revisions 312210 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r312210 | alecdavis | 2011-04-01 21:47:29 +1300 (Fri, 01 Apr 2011) | 29 lines
Merged revisions 312174 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r312174 | alecdavis | 2011-04-01 21:29:49 +1300 (Fri, 01 Apr 2011) | 23 lines
voicemail: get real last_message_index and count_messages, ODBC resequence
change last_message_index to read the max msgnum stored in the database
change count_messages to actually count the number of messages.
last_message_index change:
This fixed overwriting of the last message if msgnum=0 was missing.
Previously every incoming message would overwrite msgnum=1.
count_messages change:
allows us to detect when requencing is required in opneA_mailbox.
resequence enabled for ODBC storage:
Assists with fixing up corrupt databases with gaps, but only when
a user actively opens there mailboxes.
(closes issue #18692,#18582,#19032)
Reported by: elguero
Patches:
based on odbc_resequence_mailbox2.1.diff uploaded by elguero (license 37)
Tested by: elguero, nivek, alecdavis
Review: https://reviewboard.asterisk.org/r/1153/
........
................
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r312117 | alecdavis | 2011-04-01 20:32:12 +1300 (Fri, 01 Apr 2011) | 29 lines
Merged revisions 312103 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r312103 | alecdavis | 2011-04-01 20:25:54 +1300 (Fri, 01 Apr 2011) | 22 lines
Merged revisions 312070 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r312070 | alecdavis | 2011-04-01 19:46:56 +1300 (Fri, 01 Apr 2011) | 16 lines
app_voicemail: close_mailbox needs to respect additional messages while mailbox is open.
close_mailbox leave gaps in message sequence if messages are deleted and new messages
arrive during this time, this is because the shuffle down to slot 0, only shuffles
the number of pre-existing messages when mailbox is opened, ignoring new arrivals.
Fix: in close_mailbox re-evaluate number of messages before the shuffle, this then includes new arrivals.
Happens on filebased or ODBC storage.
(issues #19032,#18582,#18692,#18998)
Reported by: alecdavis,tootai,afosorio
Review: https://reviewboard.asterisk.org/r/1153/
........
................
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312118 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r312022 | rmudgett | 2011-03-31 15:11:40 -0500 (Thu, 31 Mar 2011) | 14 lines
chan_misdn segfaults when DEBUG_THREADS is enabled.
The segfault happens because jb->mutexjb is uninitialized from the
ast_malloc(). The internals of ast_mutex_init() were assuming a nonzero
value meant mutex tracking initialization had already happened. Recent
changes to mutex tracking code to reduce excessive memory consumption
exposed this uninitialized value.
Converted misdn_jb_init() to use ast_calloc() instead of ast_malloc().
Also eliminated redundant zero initialization code in the routine.
(closes issue #18975)
Reported by: irroot
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312023 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r311612 | bbryant | 2011-03-23 17:45:46 -0400 (Wed, 23 Mar 2011) | 9 lines
Fix a possible crash in sip/reqresp_parser.c that is caused by a possible null
value.
(closes issue #18821)
Reported by: cmaj
Patches:
patch-reqresp_parser_sip_uri_domain_cmp_c_locale-crash-1.8.3-rc2.diff.tx
uploaded by cmaj (license 830)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r311558 | twilson | 2011-03-22 19:24:53 -0700 (Tue, 22 Mar 2011) | 5 lines
Don't use static declared buf in parse_name_andor_addr
This function isn't used anywhere yet, but we definitely don't want
to keep the same value for buf between calls to the function.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r311352 | jrose | 2011-03-18 11:19:05 -0500 (Fri, 18 Mar 2011) | 10 lines
Changes some print statements/events to use a blank string in place of NULL if the string in question is NULL.
This is supposed to improve Solaris compatibility since Solaris goes berserk when trying to output NULL strings.
(closes issue #18759)
Reported by: bklang
Patches:
null-strings.patch uploaded by bklang (license 919)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311373 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r311297 | rmudgett | 2011-03-17 21:59:05 -0500 (Thu, 17 Mar 2011) | 12 lines
Race condition when ISDN CallRerouting/CallDeflection invoked.
The queued AST_CONTROL_BUSY could sometimes be processed before the
call_forward dial string is recognized.
* Moved setting the call_forwarding dial string after sending a response
to the initiator and just queue an empty frame to wake up the media thread
instead of an AST_CONTROL_BUSY.
* Added check for empty rerouting/deflection number and respond with an
error.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311298 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r311295 | rmudgett | 2011-03-17 21:22:07 -0500 (Thu, 17 Mar 2011) | 35 lines
Merged revision 310986 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
..........
r310986 | rmudgett | 2011-03-16 13:56:28 -0500 (Wed, 16 Mar 2011) | 28 lines
Dial() o option broke when connected line feature added.
The patch restores the o option behavior and adds the ability to specify
the CallerID. The Dial o and f options are complementary to each other.
The o option stores the CallerID on the outgoing channel as the channel's
CallerID. The f option forces the CallerID sent by the outgoing channel.
o(x) - The argument 'x' is optional. If not present, then specify that
the CallerID that was present on the *calling* channel be stored as the
CallerID on the *called* channel. This was the behavior of Asterisk 1.0
and earlier. If present, then specify the CallerID stored on the *called*
channel. Note that o(${CALLERID(all)}) is similar to option o without
parameters.
f(x) - The argument 'x' is optional and its presence changes the behavior
of this option. If not present, then force the outgoing CallerID on a
call-forward or deflection to the dialplan extension for this Dial() using
a dialplan 'hint'. For example, some PSTNs do not allow CallerID to be
set to anything other than the numbers assigned to you. If present, then
force the outgoing CallerID to 'x'.
Patches:
jira_abe_2752_dial_fo_options.patch uploaded by rmudgett (license 664)
Tested by: rmudgett
JIRA ABE-2752
JIRA SWP-3096
..........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311296 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r311197 | jrose | 2011-03-17 14:03:34 -0500 (Thu, 17 Mar 2011) | 11 lines
This fixes a nasty chanspy bug which was causing a channel leak every time a spied on channel made a call.
In addition to the above, it makes certain channel destruction occurs so that applications don't get stuck waiting for datastore destruction while monitored by chanspy.
(closes issue #18742)
Reported by: jkister
Tested by: jkister, jcovert, jrose
Review: http://reviewboard.digium.internal/r/106/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311198 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Richard Mudgett
Alexandr Anikin
Jonathan Rose
Leif Madsen
Alec L Davis
Matthew Nicholson
Tilghman Lesher
Russell Bryant
Brett Bryant
Terry Wilson
David Vossel