After dahdi_hangup() has supposedly hungup an ISDN channel there is still
traffic on the S0-bus because the channel was not cleaned up fully.
Shuffled the hangup code to include some missing cleanup. Also fixed some
code formatting in the area. I think the primary missing clean up code
was the call to tone_zone_play_tone() to turn off any active tones on the
channel.
(closes issue #19188)
Reported by: jg1234
Patches:
issue19188_v1.8.patch uploaded by rmudgett (license 664)
Tested by: jg1234
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@316224 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This has already been discussed and should have been resolved earlier. View
revsion 285565's log for more information about why it is important to not
put timer in the Require header.
(closes issue #18704)
Reported by: mfrager
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@316217 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r315893 | mnicholson | 2011-04-27 14:03:05 -0500 (Wed, 27 Apr 2011) | 21 lines
Merged revisions 315891 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r315891 | mnicholson | 2011-04-27 13:57:56 -0500 (Wed, 27 Apr 2011) | 14 lines
Fix our compliance with RFC 3261 section 18.2.2.
This change optimizes the free_via() function and removes some redundant null
checking. It also fixes compliance with RFC 3261 section 18.2.2 by always using
the port specified in the Via header for routing responses (even when maddr is
not set). Also the htons() function is now used when setting the port.
Additional documentation comments have been added in various places to make the
logic in the code clearer.
(closes issue #18951)
Reported by: jmls
Patches:
issue18951_set_proper_port_from_via.patch uploaded by wdoekes (license 717) (modified)
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r315672 | twilson | 2011-04-26 15:52:25 -0700 (Tue, 26 Apr 2011) | 18 lines
Merged revisions 315671 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r315671 | twilson | 2011-04-26 15:47:56 -0700 (Tue, 26 Apr 2011) | 11 lines
Make sure unregistering a peer unlinks it from the peer container
Instead of mostly copying the code from expire_register, just use the function
that "does the right thing".
(closes issue #16033)
Reported by: kkm
Patches:
016033-tilgman-fixed-refcount.diff uploaded by kkm (license 888)
Tested by: kkm, tilghman, twilson
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The 'e' extension is a fall back for the 'i', 't', or 'T' extensions if
any of them do not exist. Many of the places the 'e' extension was
supposed to be invoked fail because the priority was set wrong. There
were two places where the 'e' extension was not even checked for fall
back.
* Made invoke the 'e' extension similarly to the previous 'i', 't', or 'T'
extension check and added the 'e' extension as a fall back to the two
missing locations.
* Prioritized and optimized some hangup tests associated with the 'e'
extension.
(closes issue #19136)
Reported by: kshumard
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1196/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@315645 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r315643 | twilson | 2011-04-26 14:27:44 -0700 (Tue, 26 Apr 2011) | 25 lines
Merged revisions 315596 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r315596 | twilson | 2011-04-26 14:16:10 -0700 (Tue, 26 Apr 2011) | 18 lines
Allow transfer loops without allowing forwarding loops
We try to avoid the situation where two phones may be forwarded to each other
causing an infinite loop by storing each dialed interface in a channel
datastore and checking the list before dialing out. This works, but currently
breaks situations like A calls B, A transfers B to C, B transfers C to A, and A
transfers C to B. Since human interaction is happening here and not an
automated forwarding loop, it should be allowed.
This patch removes the dialed_interfaces datastore when a call is bridged (a
suggestion from the brilliant mmichelson). If a call is being bridged, it
should be safe to assume that we aren't stuck in a loop.
Since we are now handling this is the bridge code, the previous attempts at
handling it in app_dial and app_queue are removed.
Review: https://reviewboard.asterisk.org/r/1195/
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r315502 | tilghman | 2011-04-26 14:22:52 -0500 (Tue, 26 Apr 2011) | 21 lines
Merged revisions 315501 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r315501 | tilghman | 2011-04-26 14:18:46 -0500 (Tue, 26 Apr 2011) | 14 lines
Fix the bounds-checking code.
The code that set the bit within the select bitfield was correct, but the
bounds-checking code was not. The change to that line uses the new _bitsize
macro for clarity. Also, FD_ZERO macro did not zero-out anything but the
first word of the bitfield, so this could have caused problems with modules
using that macro with the expanded bitfield.
(closes issue #18773)
Reported by: jamicque
Patches:
20110423__issue18773.diff.txt uploaded by tilghman (license 14)
Tested by: chris-mac
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@315503 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch resolves a fairly complex deadlock that can occur with the
combination of chan_local and a dialplan switch, such as dynamic realtime
extensions, which pulls autoservice into the picture when doing a dialplan
lookup.
(closes issue #18818)
Reported by: nic
Patches:
issue18818.patch uploaded by jthurman (license 614)
18818.v1.txt uploaded by russell (license 2)
Tested by: nic, jthurman, kterzi, steve-howes, sysreq, IshMalik
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@315446 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r314778 | russell | 2011-04-22 08:58:03 -0500 (Fri, 22 Apr 2011) | 11 lines
Initialize buffers in getvar and getvarfull.
Initialize the buffers used to hold the result from GET VARIABLE or
GET VARIABLE FULL. The bug report shows func_read returning garbage in
the result. It assumed that the buffer passed in was initialized, like many
other functions do. In the more common code path (through the dialplan), it
is initialized, so just initialize it here too.
(closes issue #19050)
Reported by: johnz
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r314776 | russell | 2011-04-22 08:35:22 -0500 (Fri, 22 Apr 2011) | 10 lines
Fix handling of some call parking config options.
This patch adjusts the handling of some call parking config options to fix some
issues that have already been addressed in 1.8 and trunk.
(closes issue #19167)
Reported by: bluecrow76
Patches:
asterisk-1.6.2.17.2-fix-build-parkinglot-parked-AST_FEATURE_FLAGS.diff uploaded by bluecrow76 (license 270)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@314777 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r314620 | mnicholson | 2011-04-21 13:22:19 -0500 (Thu, 21 Apr 2011) | 20 lines
Merged revisions 314607 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r314607 | mnicholson | 2011-04-21 13:19:21 -0500 (Thu, 21 Apr 2011) | 14 lines
Added limits to the number of unauthenticated sessions TCP based protocols are allowed to have open simultaneously. Also added timeouts for unauthenticated sessions where it made sense to do so.
Unrelated, the manager interface now properly checks if the user has the "system" privilege before executing shell commands via the Originate action.
AST-2011-005
AST-2011-006
(closes issue #18787)
Reported by: kobaz
(related to issue #18996)
Reported by: tzafrir
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r314549 | twilson | 2011-04-20 17:17:34 -0700 (Wed, 20 Apr 2011) | 6 lines
Don't allocate more space than necessary for a sip_pkt
This extra allocation is a hold-over from when pkt->data was a
character array. Now that it is an allocated string, just allocate
enough for the sip_pkt.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@314550 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Return correct status: SUCCESS/FAILED/HANGUP. Previously, abnormal
exits from the command loop such as hangup would return SUCCESS.
* The "asyncagi break" command now returns SUCCESS and is now the only way
to break the command loop with that status. Previously, it returned
FAILED.
* The AMI event AsyncAGI End is no longer sent if the AsyncAGI Start event
is not sent. Previously, this happened because of an error setting up the
AGI pipes.
* All executed AGI commands now get an AsyncAGI Exec result event.
Previously, if the command returned failure (because of hangup), the
command loop just exited with FAILURE and did not send the AsyncAGI Exec
result event.
* Makes sure that the channel frame queue is empty on hangup.
Review: https://reviewboard.asterisk.org/r/1183/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@314069 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Deadlock avoidance between the sip pvt and the pvt->owner is
very difficult. Now that channel's are ao2 objects, this complication
is no longer necessary. It turns out the pvt's msg queue only
exists because of deadlock avoidance (when deadlock avoidance fails
msgs were added to a queue to be processed later), so this goes away as well.
The technique used in the new sip_lock_pvt_full() function should
be used as a template for replacing all locations where deadlock
avoidance occurs between a channel tech_pvt and the pvt's owner.
My hope is that this will begin a reversal of the invalid channel
driver locking architecture we have been using for so long.
This patch also resolves an issue where the pvt->owner gets
unlocked during processing the msg queue.
(closes issue #18690)
Reported by: dvossel
Review: https://reviewboard.asterisk.org/r/1182/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@314067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes how chan_sip handles dynamic rtp payload types
it does not understand. At the moment if a dynamic payload's mime
type does not match one we understand, the payload does not get
removed from our payload table. As a result of this, the payload
is set to whatever dynamic codec we use internally for that payload
number on outgoing INVITES. This is incorrect.
This patch fixes this by properly checking the rtpmap set function's
return code to make sure it was found. The function can return both
-1 and -2 depending on the source of the mismatch. We were just
checking -1 explicitly.
Review: https://reviewboard.asterisk.org/r/1169/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@314017 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Executing Dial(DAHDI/1/18475551212,300,) with the echotraining config
option enabled outputs the following debug messages unconditionally:
Dialing T1847555121 on 1
Dialing www2w on 1
* Made debug messages in my_dial_digits() normal debug messages that do
not get output unless enabled.
* Reworded some debug messages in my_dial_digits() to be clearer.
* Replace strncpy() with ast_copy_string() in my_dial_digits() which does
the same job better.
(closes issue #18847)
Reported by: vmikhelson
Tested by: rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@313780 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It looks like it was intentional to leave any commands or in-flight
commands in the queue in case Async AGI is run again on the call.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@313700 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r313579 | rmudgett | 2011-04-13 11:29:49 -0500 (Wed, 13 Apr 2011) | 48 lines
Merged revisions 313545 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r313545 | rmudgett | 2011-04-13 11:21:24 -0500 (Wed, 13 Apr 2011) | 41 lines
Asterisk does not hangup a channel after endpoint hangs up.
If the call that the dialplan started an AGI script for is hungup while
the AGI script is in the middle of a command then the AGI script is not
notified of the hangup. There are many AGI Exec commands that this can
happen with. The reported applications have been: Background, Wait, Read,
and Dial. Also the AGI Get Data command.
* Don't wait on the Asterisk channel after it has hung up. The channel is
likely to never need servicing again.
* Restored the AGI script's ability to return the AGI_RESULT_HANGUP value
in run_agi(). It previously only could return AGI_RESULT_SUCCESS or
AGI_RESULT_FAILURE after the DeadAGI and AGI applications were merged.
(closes issue #17954)
Reported by: mn3250
Patches:
issue17954_v1.8.patch uploaded by rmudgett (license 664)
issue17954_v1.6.2.patch uploaded by rmudgett (license 664)
issue17954_v1.4.patch uploaded by rmudgett (license 664)
Tested by: rmudgett
JIRA SWP-2171
(closes issue #18492)
Reported by: devmod
Tested by: rmudgett
JIRA SWP-2761
(closes issue #18935)
Reported by: nvitaly
Tested by: astmiv, rmudgett
JIRA SWP-3216
(closes issue #17393)
Reported by: siby
Tested by: rmudgett
JIRA SWP-2727
Review: https://reviewboard.asterisk.org/r/1165/
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* 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/branches/1.8@313517 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Merged revisions 313435 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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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/
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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/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@313434 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
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