This looks like it was just missed during a merge.
(closes issue #15841)
Reported by: amorsen
Patches:
ast_devstate_aggregate_init-in-ast_extension_state2.patch uploaded by amorsen (license 676)
Tested by: amorsen
(closes issue #15852)
Reported by: amorsen
Tested by: amorsen, farisraouf
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The variable index used in this scenario for accessing the dahdi_pvts was
wrong and was most likely copied from the several other places it is used
correctly.
(closes issue #15998)
Reported by: tsearle
Patches:
dahdi_reset_crash.patch uploaded by tsearle (license 373)
Modified:
branches/1.4/channels/chan_dahdi.c
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r222351 | jpeeler | 2009-10-06 15:35:19 -0500 (Tue, 06 Oct 2009) | 9 lines
Fix 222298 (crash during destruction of second channel when variable set with
setvar).
I mistakenly reasoned that setvar would be used on all channels. Since it can
be set per channel, give each dahdi channel a copy of the variable.
(related to #15899)
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r222298 | jpeeler | 2009-10-06 14:24:59 -0500 (Tue, 06 Oct 2009) | 9 lines
Fix crash during destruction of second channel when variable set with setvar.
The setvar line in chan_dahdi.conf is shared among all the channels, so make
sure to only free the resources only when the last channel is destroyed.
(closes issue #15899)
Reported by: tzafrir
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r222273 | tilghman | 2009-10-06 14:17:11 -0500 (Tue, 06 Oct 2009) | 5 lines
When we call a gosub routine, the variables should be scoped to avoid contaminating the caller.
This affected the ~~EXTEN~~ hack, where a subroutine might have changed the
value before it was used in the caller.
Patch by myself, tested by ebroad on #asterisk
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r222176 | kpfleming | 2009-10-05 20:24:24 -0500 (Mon, 05 Oct 2009) | 27 lines
Recorded merge of revisions 222152 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 Oct 2009) | 20 lines
Fix ao2_iterator API to hold references to containers being iterated.
See Mantis issue for details of what prompted this change.
Additional notes:
This patch changes the ao2_iterator API in two ways: F_AO2I_DONTLOCK
has become an enum instead of a macro, with a name that fits our
naming policy; also, it is now necessary to call
ao2_iterator_destroy() on any iterator that has been
created. Currently this only releases the reference to the container
being iterated, but in the future this could also release other
resources used by the iterator, if the iterator implementation changes
to use additional resources.
(closes issue #15987)
Reported by: kpfleming
Review: https://reviewboard.asterisk.org/r/383/
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r222110 | kpfleming | 2009-10-05 14:45:00 -0500 (Mon, 05 Oct 2009) | 25 lines
Allow non-compliant T.38 endpoints to be supportable via configuration option.
Many T.38 endpoints incorrectly send the maximum IFP frame size they can accept
as the T38FaxMaxDatagram value in their SDP, when in fact this value is
supposed to be the maximum UDPTL payload size (datagram size) they can accept.
If the value they supply is small enough (a commonly supplied value is '72'),
T.38 UDPTL transmissions will likely fail completely because the UDPTL packets
will not have enough room for a primary IFP frame and the redundancy used for
error correction. If this occurs, the Asterisk UDPTL stack will emit log messages
warning that data loss may occur, and that the value may need to be overridden.
This patch extends the 't38pt_udptl' configuration option in sip.conf to allow
the administrator to override the value supplied by the remote endpoint and
supply a value that allows T.38 FAX transmissions to be successful with that
endpoint. In addition, in any SIP call where the override takes effect, a debug
message will be printed to that effect. This patch also removes the
T38FaxMaxDatagram configuration option from udptl.conf.sample, since it has not
actually had any effect for a number of releases.
In addition, this patch cleans up the T.38 documentation in sip.conf.sample
(which incorrectly documented that T.38 support was passthrough only).
(issue #15586)
Reported by: globalnetinc
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r221844 | rmudgett | 2009-10-01 20:09:31 -0500 (Thu, 01 Oct 2009) | 33 lines
Merged revisions 221769 via svnmerge from
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r221769 | rmudgett | 2009-10-01 18:18:28 -0500 (Thu, 01 Oct 2009) | 26 lines
Occasionally losing use of B channels in chan_misdn.
I have not been able to reproduce the problem of losing channels.
However, I have seen in the code a reentrancy problem that might give
these symptoms.
The reentrancy patch does several things:
1) Guards B channel and B channel structure allocation.
2) Makes the B channel structure find routines more precise in locating records.
3) Never leave a B channel allocated if we received cause 44.
The last item may cause temporary outgoing call problems, but they should
clear when the line becomes idle.
(closes issue #15490)
Reported by: slutec18
Patches:
issue15490_channel_alloc_reentrancy.patch uploaded by rmudgett (license 664)
Tested by: rmudgett, slutec18
(closes issue #15458)
Reported by: FabienToune
Patches:
issue15458_channel_alloc_reentrancy.patch uploaded by rmudgett (license 664)
Tested by: FabienToune, rmudgett, slutec18
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r221592 | kpfleming | 2009-10-01 11:16:09 -0500 (Thu, 01 Oct 2009) | 12 lines
Remove ability to control T.38 FAX error correction from udptl.conf.
chan_sip has had the ability to control T.38 FAX error correction mode on a per-peer
(or global) basis for a couple of releases now, which is where it should have been
all along. This patch removes the ability to configure it in udptl.conf, but issues
a warning if the user tries to do, telling them to look at sip.conf.sample for how
to configure it now. For any SIP peers that are T.38 enabled in sip.conf, there is
already a default for FEC error correction even if the user does not specify any mode,
so this change will not turn off error correction by default, it will have the same
default value that has been in the udptl.conf sample file.
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r221432 | mnicholson | 2009-09-30 15:40:20 -0500 (Wed, 30 Sep 2009) | 17 lines
Merged revisions 221360 via svnmerge from
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r221360 | mnicholson | 2009-09-30 14:36:06 -0500 (Wed, 30 Sep 2009) | 10 lines
Fix SRV lookup and Request-URI generation in chan_sip.
This patch adds a new field "portinuri" to the sip dialog struct and the sip peer struct. That field is used during RURI generation to determine if the port should be included in the RURI. It is also used in some places to determine if an SRV lookup should occur.
(closes issue #14418)
Reported by: klaus3000
Tested by: klaus3000, mnicholson
Review: https://reviewboard.asterisk.org/r/369/
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r221266 | twilson | 2009-09-30 12:52:30 -0500 (Wed, 30 Sep 2009) | 32 lines
Merged revisions 221086 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) | 25 lines
Change the SSRC by default when our media stream changes
Be default, change SSRC when doing an audio stream changes Asterisk doesn't
honor marker bit when reinvited to already-bridged RTP streams,resulting in
far-end stack discarding packets with "old" timestamps that areactually part of
a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is a
reinvite, unless the 'constantssrc' is set to true in sip.conf.
The original issue reported to Digium support detailed the following situation:
ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in
fromITSP, Asterisk dials the app server which sends a re-invite back
toAsterisk--not to negotiate to send media directly to the ITSP, but to
indicatethat it's changing the stream it's sending to Asterisk. The app
servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker
bit on the new stream. Asterisk passes through the teimstamp of the new stream,
butdoes not reset the SSRC, sequence numbers, or set the marker bit.
When the timestamp on the new stream is older than the timestamp on the
originalstream, the ITSP (which doesn't know there has been any change) discards
the newframes because it thinks they are too old. This patch addresses this by
changing the SSRC on a stream update unless constantssrc=true is set in
sip.conf.
Review: https://reviewboard.asterisk.org/r/374/
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r220906 | tilghman | 2009-09-29 14:57:37 -0500 (Tue, 29 Sep 2009) | 16 lines
Merged revisions 220873 via svnmerge from
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r220873 | tilghman | 2009-09-29 12:59:26 -0500 (Tue, 29 Sep 2009) | 9 lines
Reduce CPU usage related to building a peer merely for devicestates.
This fixes a 100% CPU problem in the SIP driver, found by profiling
the driver while the problem was occurring.
(closes issue #14309)
Reported by: pkempgen
Patches:
20090924__issue14309.diff.txt uploaded by tilghman (license 14)
Tested by: pkempgen, vrban
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r220833 | jpeeler | 2009-09-29 11:58:29 -0500 (Tue, 29 Sep 2009) | 12 lines
Make deletion of temporary greetings work properly with IMAP_STORAGE
When imapgreetings was set to yes, the message was being deleted but wasn't
actually being expunged. When imapgreetings was set to no, the file based
message was not being deleted at all. All good now!
(closes issue #14949)
Reported by: noahisaac
Patches:
vm_tempgreeting_removal.patch uploaded by noahisaac (license 748),
modified by me
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r220718 | jpeeler | 2009-09-28 14:10:10 -0500 (Mon, 28 Sep 2009) | 10 lines
Fix building of registration entry in build_peer when using callbackextension
Check for remotesecret option was unintentionally always true, which therefore
caused the secret option to never be used. Thanks to dvossel for pointing out
the exact fix.
(closes issue #15943)
Reported by: tpsast
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r220629 | mvanbaak | 2009-09-27 22:40:16 +0200 (Sun, 27 Sep 2009) | 3 lines
add name argument for the CALLERID dialplan function to the xml documentation.
Pointed out to me on IRC by snuff-home. Thanks
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r220028 | mvanbaak | 2009-09-24 10:36:18 +0200 (Thu, 24 Sep 2009) | 14 lines
Merged revisions 220027 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r220027 | mvanbaak | 2009-09-24 10:33:50 +0200 (Thu, 24 Sep 2009) | 7 lines
mkpkgconfig does not need bash so make it use /bin/sh
This fixes building on all systems that don't have bash
at /bin/bash
Reported by _ys on #asterisk-dev
Tested by _ys on #asterisk-dev
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r219895 | lmadsen | 2009-09-23 12:46:46 -0500 (Wed, 23 Sep 2009) | 13 lines
Add Mantis work flow documention.
This commit adds the doxygen changes that I've made to describe the Mantis
work flow documentation for the open source issue tracker. This should make
it easier to determine the flow of issues through the issue tracker, and what
those statuses mean.
(closes issue #15902)
Reported by: lmadsen
Patches:
mantisworkflow.h uploaded by lmadsen (license 10)
Review: https://reviewboard.asterisk.org/r/367/
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r219818 | tilghman | 2009-09-22 16:43:22 -0500 (Tue, 22 Sep 2009) | 17 lines
Merged revisions 219816 via svnmerge from
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r219816 | tilghman | 2009-09-22 16:37:03 -0500 (Tue, 22 Sep 2009) | 10 lines
When IMAP variables were changed during a reload, Voicemail did not use the new values.
This change introduces a configuration version variable, which ensures that
connections with the old values are not reused but are allowed to expire
normally.
(closes issue #15934)
Reported by: viniciusfontes
Patches:
20090922__issue15934.diff.txt uploaded by tilghman (license 14)
Tested by: viniciusfontes
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r219520 | dvossel | 2009-09-18 18:20:58 -0500 (Fri, 18 Sep 2009) | 15 lines
Merged revisions 219519 via svnmerge from
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r219519 | dvossel | 2009-09-18 18:19:50 -0500 (Fri, 18 Sep 2009) | 9 lines
iax2 frame double free
The iax frame's retrans sched id was written over right
before iax2_frame_free was called. In iax2_frame_free that
retrans id is used to delete the sched item. By writing over
the retrans field before the sched item could be deleted, it was
possible for a retransmit to occur on a freed frame.
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r219451 | dvossel | 2009-09-18 11:20:41 -0500 (Fri, 18 Sep 2009) | 20 lines
Merged revisions 219450 via svnmerge from
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r219450 | dvossel | 2009-09-18 11:19:15 -0500 (Fri, 18 Sep 2009) | 14 lines
via-header branches not updated correctly on INVITE
INVITE requests must always contain a new unique branch id. When
a new branch id is created for an INVITE, the dialog's invite_branch
variable must be updated so CANCEL requests use the correct branch id.
(closes issue #15262)
Reported by: maniax
Patches:
asterisk-1.6.1.0-sip-branch.patch uploaded by tweety (license 608)
invite_new_branch_trunk.diff uploaded by dvossel (license 671)
Tested by: maniax, dvossel
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