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The first real test added to the external test suite found a pretty nasty crash
that occurred in Asterisk trunk. The crash was due to a race condition between
the REFER handling and channel destruction in the channel thread. After the
transfer has been completed, we go back to the transferrer channel and try to
lock it so we can fire off a CEL event. However, there was no guarantee that
the channel was still around at that point since it's racing against the channel
thread.
Since ast_channel is a reference counted object, the fix is simple. The code
unlocks the transferrer channel before finally completing the transfer with
an async goto. At this point the channel thread is going to start call tear
down and the channel will eventually be destroyed. To ensure that the channel
is valid when we want to fire off the CEL event, increase the channel's
reference count.
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The PITCH_SHIFT function can be used on a channel to independently
modify the pitch of both rx and tx audio streams. Now you can
improve your conference calls by assigning a random pitch effect
to everyone entering a meetme room, or just make your day more
interesting by making your co-workers sound funny. These are just
some of the numerious practical uses for this function. Enjoy!
https://reviewboard.asterisk.org/r/526/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@251038 f38db490-d61c-443f-a65b-d21fe96a405b
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Review: https://reviewboard.asterisk.org/r/541/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@251022 f38db490-d61c-443f-a65b-d21fe96a405b
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Fixes regression introduced in 140167 that uses the wrong variable names.
(closes issue #16930)
Reported by: ianc
Patches:
fix_reload_followme.diff uploaded by ianc (license 998)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250979 f38db490-d61c-443f-a65b-d21fe96a405b
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The get_local_address() function for an RTP instance was used when building an
SDP, but the results were not honored. The RTP engine activate() function was
not being used once we have determined that media will now flow.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250917 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16953)
Reported by: elguero
Patches:
app_voicemail-odbc-syntax-fix.diff uploaded by elguero (license 37)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250913 f38db490-d61c-443f-a65b-d21fe96a405b
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The mis-placement of the latest entry meant that when it was set, it was writing
one index past the end of the properties array in the ast_rtp_instance (which
happened to be the local_address field).
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r250786 | jpeeler | 2010-03-04 19:02:58 -0600 (Thu, 04 Mar 2010) | 9 lines
Fix not being able to specify a URL in MOH class directory.
Don't attempt to chdir on a URL!
(closes issue #16875)
Reported by: raarts
Patches:
moh-http.patch uploaded by raarts (license 937)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r250613 | lmadsen | 2010-03-03 16:28:02 -0500 (Wed, 03 Mar 2010) | 11 lines
Update existing Local channel documentation.
A complete re-write of the Local channel documentation has been performed, with
the existing information from localchannel.txt and localchannel.tex merged in.
(issue #16637)
Reported by: kobaz
Patches:
localchannel.tex uploaded by lmadsen (license 10)
localchannel.txt uploaded by lmadsen (license 10)
Tested by: lmadsen, jsmith, mmichelson
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A complete re-write of the Local channel documentation has been performed, with
the existing information from localchannel.txt and localchannel.tex merged in.
(closes issue #16637)
Reported by: kobaz
Patches:
localchannel.tex uploaded by lmadsen (license 10)
localchannel.txt uploaded by lmadsen (license 10)
Tested by: lmadsen, jsmith, mmichelson
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250609 f38db490-d61c-443f-a65b-d21fe96a405b
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Only chan_dahdi set a value in cdrflags. Everyone else just copied it
around the system. Noone cared about any value it may have contained.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250565 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r250480 | jpeeler | 2010-03-03 13:04:11 -0600 (Wed, 03 Mar 2010) | 15 lines
Make sure to clear red alarm after polarity reversal.
From the issue:
The automatic overnight line tests (or manual ones) used on UK (BT) lines causes
a red alarm on a dahdi / TDM400P connected channel. This is because the line
uses voltage tests (battery loss) and polarity reversal. The polarity reversal
causes chan_dahdi to initiate v23 CallerID processing but during this the event
DAHDI_EVENT_NOALARM is ignored so that the alarm is never cleared.
(closes issue #14163)
Reported by: jedi98
Patches:
chan_dahdi-1.4-inalarm.diff uploaded by jedi98 (license 653)
Tested by: mattbrown, Chainsaw, mikeeccleston
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r250394 | dvossel | 2010-03-03 12:02:27 -0600 (Wed, 03 Mar 2010) | 16 lines
fixes problem with duplicate TXREQ packets
When Asterisk receives an IAX2 TXREQ packet, try_transfer()
will call store_by_transfercallno() to link the chan_iax2_pvt
struct into iax_transfercallno_pvts. If a duplicate TXREQ
packet is received for the same call, the pvt struct will be
linked into iax_transfercallno_pvts multiple times. This patch
fixes this. Thanks rain for debugging this and providing a patch!
(closes issue #16904)
Reported by: rain
Patches:
iax2-double-txreq-fix.diff uploaded by rain (license 327)
Tested by: rain, dvossel
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250395 f38db490-d61c-443f-a65b-d21fe96a405b
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New config parameter "reportalarms" added in chan_dahdi.conf which supports the
following possible values:
"channels": report each channel alarms (current behavior, default for backward compatibility)
"spans": report an "SpanAlarm" event when the span of any configured channel is alarmed
"all": report channel and span alarms (aggregated behavior)
"none": do not report any alarms
(closes issue #16709)
Reported by: nahuelgreco
Patches:
chan_dahdi.c.reportalarms.patch uploaded by nahuelgreco (license 162)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250392 f38db490-d61c-443f-a65b-d21fe96a405b
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Also fixed MODULEINFO depends and conflicts for app_fax, res_fax, and res_fax_spandsp.
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The 'p' option allows the PickupChan app to pickup
a ringing phone by looking for the first match to a
partial channel name rather than requiring a full match.
(closes issue #16613)
Reported by: syspert
Patches:
pickipbycallid.patch uploaded by syspert (license 938)
pickupbycallerid_v2.patch uploaded by dvossel (license 671)
Tested by: dvossel, syspert
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250141 f38db490-d61c-443f-a65b-d21fe96a405b
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Update the IMAP documentation to make it clear that storing voicemails
in the same folder as a large number of emails could potentially cause
significant slow downs when writing or retrieving voicemails.
(issue #16704)
Reported by: TimeHider
Tested by: lmadsen, TimeHider
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250051 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r250043 | lmadsen | 2010-03-02 15:51:35 -0500 (Tue, 02 Mar 2010) | 7 lines
Update documentation to clarify purpose of unanswered option.
(closes issue #16267)
Reported by: elsto
Patches:
cdr.conf.sample.patch.txt uploaded by lmadsen (license 10)
Tested by: davidw, elsto
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r250041 | lmadsen | 2010-03-02 15:45:37 -0500 (Tue, 02 Mar 2010) | 4 lines
Update documentation to not imply we support overriding options.
(issue #16855)
Reported by: davidw
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(closes issue #16855)
Reported by: davidw
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(closes issue #16145)
Reported by: tilghman
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caused a regression, as only supported VOICE, not VIDEO etc.
(issue #16880)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249947 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16743)
Reported by: snuffy
Patches:
parkingdoc.diff uploaded by snuffy (license 35)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249925 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16798)
Reported by: snuffy
Patches:
xml_dundi.diff uploaded by snuffy (license 35)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249912 f38db490-d61c-443f-a65b-d21fe96a405b
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When configuring the adaptive jitterbuffer, the target_extra
value not only could not be set from the configuration, but was
not even being set to its proper default. This value is required
in order for the adaptive jitterbuffer to work correctly. To resolve
this a config option has been added to expose this value to the conf
files, and a default value is provided when no config specific value
is present.
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VMSayName that will play the recorded name of the voicemail user if it exists,
otherwise will play the mailbox number. A unit test has been written to verify
correct functionality called test_voicemail_vmsayname.
(closes issue #14973)
Reported by: ghjm
Review: https://reviewboard.asterisk.org/r/530/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249889 f38db490-d61c-443f-a65b-d21fe96a405b
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when called from a ISDN channel, null frames prevent '#' exit.
Now only echo back VOICE and DTMF frames
(issue #16880)
Reported by: alecdavis
Patches:
echo_exit.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249801 f38db490-d61c-443f-a65b-d21fe96a405b
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regression since sig_pri split.
(issue #16909)
Reported by: alecdavis
Patches:
pritimer.asterisk.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249759 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r249671 | seanbright | 2010-03-01 14:35:01 -0500 (Mon, 01 Mar 2010) | 11 lines
Fix crash in app_voicemail related to message counting.
We were passing a 'struct inprocess **' and treating it like a 'struct inprocess *'
causing a segfault.
(closes issue #16921)
Reported by: whardier
Patches:
20100301_issue16921.patch uploaded by seanbright (license 71)
Tested by: whardier
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This patch splits transmit_displaymessage into transmit_clear_display_message and transmit_display_message which better aligns with the skinny protocol. The new transmit_display_message is not used in the current code, but will be and so it is commented.
Moved handle_datetime from this function to onhook and offhook functions (display now properly cleared at the end of a call on 30VIP).
Removed skinny debug messages from inline code as there's an ast_verb in transmit_clear_display_message. Also, removed commentary that it was a clear display as it is now apparent from the function name.
Split transmit_displaypromptmessage into display and clear.
(closes issue #16878)
Reported by: wedhorn
Patches:
skinny-clean02.diff uploaded by wedhorn (license 30)
skinny-clean03.diff uploaded by wedhorn (license 30)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249670 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16826)
Reported by: PipoCanaja
Patches:
chan_skinny.c_bigendianPatch_20100218.diff uploaded by PipoCanaja (license 994)
Tested by: wedhorn
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r249536 | jpeeler | 2010-03-01 11:02:03 -0600 (Mon, 01 Mar 2010) | 11 lines
Modify queued frames from local channels to not set the other side to up
In this case, attended transfers were broken due to ast_feature_request_and_dial
detecting the channel being set to up before the answer frame could be read and
therefore failing to mark the channel as ready. This fix is a regression fix for
244785, which should continue to work properly as well.
(closes issue #16816)
Reported by: jamhed
Tested by: jamhed, corruptor
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(closes issue #16927)
Reported by: alecdavis
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249491 f38db490-d61c-443f-a65b-d21fe96a405b
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previous test, gave false level of assurance that code was healthy.
(issue #16927)
Reported by: alecdavis
Patches:
based on app_voicemail_test.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249449 f38db490-d61c-443f-a65b-d21fe96a405b
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the -dev list.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249405 f38db490-d61c-443f-a65b-d21fe96a405b
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Following Q.931 5.2.4
When the user has determined that sufficient call information has been received the
user shall stop T302 and send CALL PROCEEDING to the network.
Previously timeouts were possible if the dialplan took a long time to issue any
response back to the network.
Verified that our local TELCO also does the same.
(issue #16789)
Reported by: alecdavis
Patches:
overlap_receiving_trunk.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249320 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r249234 | kpfleming | 2010-02-27 09:07:59 -0500 (Sat, 27 Feb 2010) | 1 line
add a reference to the now-published IAX2 RFC
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