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Don't always define HAVE_ZAPTEL_CHANALARMS (since we check if it's defined..)
Minor cleanup to make things clear.
(closes issue #13726)
Reported by: tzafrir
Patches:
dahdi_def.diff uploaded by tzafrir (license 46)
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headers or body lines that we support
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comes from Sean Bright. Thanks Sean!
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IP/port does not change.
(closes issue #13309)
Reported by: dimas
Patches:
v2-13309.patch uploaded by dimas (license 88)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@149207 f38db490-d61c-443f-a65b-d21fe96a405b
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lines in sip.conf.
(closes issue #13570)
Reported by: putnopvut
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(closes issue #13676)
Reported by: tacvbo
Patches:
13676.patch uploaded by putnopvut (license 60)
Tested by: tacvbo
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stack is actually large enough to hold the legal values of that option, and also ensure that sscanf() knows to stop parsing if it would overrun the buffer (without these changes, specifying "buffers=...,immediate" would overflow the buffer on the stack, and could not have worked as expected)
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changes
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setting channel variables and other things on the channel, otherwise the channel might go away (if the dialplan hangs up quickly) before we are done, which results in a spectacular crash
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workspace. To signal an error (and no change to the workspace), -1 should be
returned instead.
(closes issue #13340)
Reported by: kryptolus
Patches:
20080827__bug13340__2.diff.txt uploaded by Corydon76 (license 14)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@146799 f38db490-d61c-443f-a65b-d21fe96a405b
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method, because we need to evaluate the callerid (since that data affects
whether an extension exists).
(closes issue #13343)
Reported by: efutch
Patches:
20080915__bug13343.diff.txt uploaded by Corydon76 (license 14)
Tested by: efutch
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@146711 f38db490-d61c-443f-a65b-d21fe96a405b
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'leaking' configuration data from other configured channels
(closes issue #13555)
Reported by: jeffg
Patches:
issue_13555.patch uploaded by kpfleming (license 421)
Tested by: jeffg
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channels/misdn/isdn_lib.c
* Miscellaneous other fixes from trunk to make merging easier later.
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r145200 | rmudgett | 2008-09-30 16:00:54 -0500 (Tue, 30 Sep 2008) | 7 lines
* Miscellaneous formatting changes to make v1.4 and trunk
more merge compatible in the mISDN area.
channels/chan_misdn.c
* Eliminated redundant code in cb_events() EVENT_SETUP
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r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008) | 9 lines
improved helptext of misdn_set_opt.
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r142181 | rmudgett | 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) | 1 line
Cleaned up comment
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r138738 | rmudgett | 2008-08-18 16:07:28 -0500 (Mon, 18 Aug 2008) | 30 lines
channels/chan_misdn.c
* Made bearer2str() use allowed_bearers_array[]
* Made use the causes.h defines instead of hardcoded numbers.
* Made use Asterisk presentation indicator values if either of the
mISDN presentation or screen options are negative.
* Updated the misdn_set_opt application option descriptions.
* Renamed the awkward Caller ID presentation misdn_set_opt
application option value not_screened to restricted.
Deprecated the not_screened option value.
channels/misdn/isdn_lib.c
* Made use the causes.h defines instead of hardcoded numbers.
* Fixed some spelling errors and typos.
* Added all defined facility code strings to fac2str().
channels/misdn/isdn_lib.h
* Added doxygen comments to struct misdn_bchannel.
channels/misdn/isdn_lib_intern.h
* Added doxygen comments to struct misdn_stack.
channels/misdn_config.c
configs/misdn.conf.sample
* Updated the mISDN presentation and screen parameter descriptions.
doc/misdn.txt (doc/tex/misdn.tex)
* Updated the misdn_set_opt application option descriptions.
* Fixed some spelling errors and typos.
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Reported by: MVF
Tested by: neutrino88, urzedo, murf, thiagofernandes
Many thanks to neutrino88 for this patch, which
solves a problem whereby channels get a CANCEL
request, respond to it properly, but end up
in a hung state, infinitely being rescheduled.
This fix is a bit crude, in that catches the
problem at a rather late phase, but it may
prevent infinite rescheduling problems that
might still arise.
It might have been better to find out why,
in the course of protocol handling, the channel
was not destroyed, but we leave that to
future generations.
Many thanks to urzedo and thiagofernandes for
their work in verifying that the patch code
indeed is being executing, and averting the
problem.
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using it.
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(closes issue #13240)
Reported by: jvandal
Patches:
20080813__bug13240.diff.txt uploaded by Corydon76 (license 14)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@143140 f38db490-d61c-443f-a65b-d21fe96a405b
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Reported by: rlsutton2
Prevents local channels from playing MOH at each other which was causing ast_generic_bridge to loop much faster.
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improve the security of various installations. As this does not change
any default behavior, it is not classified as a direct security fix for
anything within Asterisk, but may help PBX admins better secure their
SIP servers.
(closes issue #11776)
Reported by: ibc
Patches:
20080829__bug11776.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76, blitzrage
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@142865 f38db490-d61c-443f-a65b-d21fe96a405b
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matches the branch of the INVITE it is cancelling.
(closes issue #13381)
Reported by: atca_pres
Patches:
13381v2.patch uploaded by putnopvut (license 60)
Tested by: atca_pres
(closes issue #13198)
Reported by: rickead2000
Tested by: rickead2000
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the media to be natively bridged, use the jointcapability
instead of the peercapability.
It seems that the intent of using the peercapability was to
expand the choice of codecs for the call to increase the
chances of being able to native bridge the channels. The
problem is that if a codec were settled on for the native
bridge and that wasn't a codec that was configured to be used
by Asterisk for that peer, then Asterisk would send a
REINVITE with no codecs in the SDP which is a bug no matter
how you slice it.
(closes issue #13076)
Reported by: ramonpeek
Patches:
13076.patch uploaded by putnopvut (license 60)
Tested by: tbelder
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@142079 f38db490-d61c-443f-a65b-d21fe96a405b
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remote tag of an endpoint once a dialog has
been confirmed. Up until that point, it is possible
and legal for the far-end to send provisional
responses with a different To: tag each time. With
this patch applied, these provisional messages
will not cause a matching problem.
(closes issue #11536)
Reported by: ibc
Patches:
11536v2.patch uploaded by putnopvut (license 60)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@141809 f38db490-d61c-443f-a65b-d21fe96a405b
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with a solution. This will most likely will close 13235 and 13409. I'll wait till Monday to verify, and then close these bugs.
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if the channel has been hung up. It will likely crash
otherwise
ABE-1159
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caused port 5060 to always be used when dialing
a peer if no explicit port was specified. This
broke the behavior of implicitly using the port
from which the peer registered if no port is
specified. This commit fixes the logic flaw.
(closes issue #13424)
Reported by: mdu113
Patches:
13424.patch uploaded by putnopvut (license 60)
Tested by: mdu113
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@141217 f38db490-d61c-443f-a65b-d21fe96a405b
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buffers when copying configuration values.
(closes issue #13336)
Reported by: decryptus_proformatique
Patches:
chan_iax2_mohinterpret_mohsuggest_general_settings.patch uploaded by decryptus (license 555)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@140605 f38db490-d61c-443f-a65b-d21fe96a405b
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something a bit strange. In all cases where we provide
a callback function to ao2_container_alloc, the callback
function would only return 0 or CMP_MATCH. After inspecting
the ao2_callback() code carefully, I found that if you're
only looking for one specific item, then you should return
CMP_MATCH | CMP_STOP. Otherwise, astobj2 will continue
traversing the current bucket until the end searching for
more matches.
In cases like chan_iax2 where in 1.4, all the peers are
shoved into a single bucket, this makes for potentially
terrible performance since the entire bucket will be
traversed even if the peer is one of the first ones come
across in the bucket.
All the changes I have made were for cases where the
callback function defined was passed to ao2_container_alloc
so that calls to ao2_find could find a unique instance
of whatever object was being stored in the container.
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in a string to Dial(), it is not ignored.
(closes issue #13355)
Reported by: acunningham
Patches:
13355v2.patch uploaded by putnopvut (license 60)
Tested by: acunningham
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@140417 f38db490-d61c-443f-a65b-d21fe96a405b
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in pedantic mode. The problem was that the wrong
tags would be compared depending on the direction
of the call.
(closes issue #13353)
Reported by: flefoll
Patches:
chan_sip.c.br14.139015.patch-refer-pedantic uploaded by flefoll (license 244)
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of a completely unrelated function to determine whether the scheduler should
be run or not. This would have caused the scheduler to not run in cases where
it should have. Also, leave a note about another scheduler issue that needs
to be addressed at some point.
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Reported by: tzafrir
Patches:
dahdi_close.diff uploaded by tzafrir (license 46)
Tested by: tzafrir, jpeeler
This patch fixes closing open file descriptors in the case of an error.
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used here to allow newly scheduled tasks to wake up the scheduler just in case
the new task needs to run sooner than the current wakeup time when the thread
is sleeping. However, there was a race condition such that a newly scheduled
task would not properly wake up the scheduler or affect the wake up period.
The order of execution would have been:
1) Scheduler thread determines wake up time of N ms.
2) Another thread schedules a task and signals the condition, with an
execution time of < N ms.
3) Scheduler thread locks and goes to sleep for N ms.
By moving the sleep time determination to inside the critical section, this
possibility is avoided.
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incorrect locking order between iax2_pvt and
ast_channel structures.
AST-13
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being able to be opened or specify after a restart.
(closes issue #11017)
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because not doing so just makes it difficult for people with channels that are in alarm when Asterisk starts up to get them going once the alarm is cleared
(closes issue #12160)
Reported by: tzafrir
Patches:
asterisk-chanalarms_14.patch uploaded by tzafrir (license 46)
Tested by: tzafrir
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@139145 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #13257)
Reported by: travishein
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(from what the corresponding channel variables were set to)
when the agent logs out.
(closes issue #13098)
Reported by: davidw
Patches:
20080731__issue13098_agent_ackcall_not_reset.diff uploaded by bbryant (license 36)
Tested by: davidw
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@138942 f38db490-d61c-443f-a65b-d21fe96a405b
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number of buffers to be configured per channel. Syntax:
buffers=<num of buffers>,<policy>
Where the number of buffers is some non-negative integer and the policy is either "full", "half", or "immediate".
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destroyed allowing module to be unloaded
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(closes issue #12921)
Reported by: Nuitari
Patches:
20080804__bug12921.diff.txt uploaded by Corydon76 (license 14)
20080815__bug12921.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@138258 f38db490-d61c-443f-a65b-d21fe96a405b
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