https://origsvn.digium.com/svn/asterisk/trunk
........
r223330 | kpfleming | 2009-10-09 15:58:44 -0500 (Fri, 09 Oct 2009) | 10 lines
Initiate T.38 switchover when acting as called party, regardless of FAX direction.
SendFAX() and ReceiveFAX() can be given options to indicate whether they should
act as the calling or called party; this mode should be used to decide whether
to initiate a switchover to T.38, not the direction that the FAX transfer will
take place.
(closes issue #16039)
Reported by: jamicque
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@223331 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r223136 | mnicholson | 2009-10-09 12:14:38 -0500 (Fri, 09 Oct 2009) | 8 lines
Don't close the sqlite database when reloading. Only close the database when unloading.
(closes issue #15953)
Reported by: frawd
Patches:
sqlite3_rev220097.diff uploaded by frawd (license 610)
Tested by: frawd
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@223172 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r223088 | dvossel | 2009-10-09 10:49:30 -0500 (Fri, 09 Oct 2009) | 14 lines
p->peerauth is always empty in transmit_register()
When using callbackextension or specifing the peer name
in a registration string, the peer's specific auth settings
set by the "auth=" strings within the peer definition are not
used by the registration. Thanks to ebroad for reporting the
issue and providing the patch.
(closes issue #15955)
Reported by: ebroad
Patches:
regauthfix.patch uploaded by ebroad (license 878)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@223091 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r222880 | russell | 2009-10-08 14:52:03 -0500 (Thu, 08 Oct 2009) | 51 lines
Merged revisions 222878 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r222878 | russell | 2009-10-08 14:45:47 -0500 (Thu, 08 Oct 2009) | 44 lines
Make filestream frame handling safer by isolating frames before returning them.
This patch is related to a number of issues on the bug tracker that show
crashes related to freeing frames that came from a filestream. A number of
fixes have been made over time while trying to figure out these problems, but
there re still people seeing the crash. (Note that some of these bug reports
include information about other problems. I am specifically addressing
the filestream frame crash here.)
I'm still not clear on what the exact problem is. However, what is _very_
clear is that we have seen quite a few problems over time related to unexpected
behavior when we try to use embedded frames as an optimization. In some cases,
this optimization doesn't really provide much due to improvements made in other
areas.
In this case, the patch modifies filestream handling such that the embedded frame
will not be returned. ast_frisolate() is used to ensure that we end up with a
completely mallocd frame. In reality, though, we will not actually have to malloc
every time. For filestreams, the frame will almost always be allocated and freed
in the same thread. That means that the thread local frame cache will be used.
So, going this route doesn't hurt.
With this patch in place, some people have reported success in not seeing the
crash anymore.
(SWP-150)
(AST-208)
(ABE-1834)
(issue #15609)
Reported by: aragon
Patches:
filestream_frisolate-1.4.diff2.txt uploaded by russell (license 2)
Tested by: aragon, russell
(closes issue #15817)
Reported by: zerohalo
Tested by: zerohalo
(closes issue #15845)
Reported by: marhbere
Review: https://reviewboard.asterisk.org/r/386/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@222881 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r222799 | rmudgett | 2009-10-08 11:44:33 -0500 (Thu, 08 Oct 2009) | 19 lines
Merged revisions 222797 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r222797 | rmudgett | 2009-10-08 11:33:06 -0500 (Thu, 08 Oct 2009) | 12 lines
Fix memory leak if chan_misdn config parameter is repeated.
Memory leak when the same config option is set more than once in an
misdn.conf section. Why must this be considered? Templates! Defining a
template with default port options and later adding to or overriding some
of them.
Patches:
memleak-misdn.patch
JIRA ABE-1998
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@222800 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r222692 | rmudgett | 2009-10-07 16:56:36 -0500 (Wed, 07 Oct 2009) | 21 lines
Merged revisions 222691 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r222691 | rmudgett | 2009-10-07 16:51:24 -0500 (Wed, 07 Oct 2009) | 14 lines
chan_misdn.c:process_ast_dsp() memory leak
misdn.conf: astdtmf must be set to "yes". With "no", buffer loss does not
occur.
The translated frame "f2" when passing through ast_dsp_process() is not
freed whenever it is not used further in process_ast_dsp(). Then in the
end it is never ever freed.
Patches:
translate.patch
JIRA ABE-1993
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@222693 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@222605 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@222395 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
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)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@222352 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@222302 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@222279 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
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
........
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/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@222185 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@222111 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r221844 | rmudgett | 2009-10-01 20:09:31 -0500 (Thu, 01 Oct 2009) | 33 lines
Merged revisions 221769 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@221853 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
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.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@221598 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r221432 | mnicholson | 2009-09-30 15:40:20 -0500 (Wed, 30 Sep 2009) | 17 lines
Merged revisions 221360 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@221486 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
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
........
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/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@221301 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r220906 | tilghman | 2009-09-29 14:57:37 -0500 (Tue, 29 Sep 2009) | 16 lines
Merged revisions 220873 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@220976 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@220834 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@220719 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@220630 65c4cc65-6c06-0410-ace0-fbb531ad65f3
