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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174325 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174301 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174046 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173974 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173776 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173697 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173693 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173502 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173500 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173458 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173354 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173249 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173130 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172890 65c4cc65-6c06-0410-ace0-fbb531ad65f3
