Age | Commit message (Collapse) | Author | Files | Lines |
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responses)
(closes issue #16202)
Reported by: jsutton
Patches:
20100504__issue16202.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
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r262414 | pabelanger | 2010-05-11 15:26:17 -0400 (Tue, 11 May 2010) | 8 lines
Improve logging information for misconfigured contexts
(closes issue #17238)
Reported by: pprindeville
Patches:
chan_sip-bug17238.patch uploaded by pprindeville (license 347)
Tested by: pprindeville
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r262236 | dvossel | 2010-05-10 13:36:10 -0500 (Mon, 10 May 2010) | 11 lines
fixes crash in chan_console
There is a race condition between console_hangup()
and start_stream(). It is possible for console_hangup()
to be called and then the stream thread to begin after the hangup.
To avoid this a check in start_stream() to make sure the pvt-owner
still exists while the pvt lock is held is made. If the owner
is gone that means the channel hung up and start_stream should
be aborted.
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r261560 | tilghman | 2010-05-06 10:39:10 -0500 (Thu, 06 May 2010) | 8 lines
Permit more lines within a SIP body to be parsed.
The example given within the related issue showed 120 lines, which was mostly
a result of the body being XML.
(closes issue #17179)
Reported by: khw
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r261314 | pabelanger | 2010-05-05 14:43:03 -0400 (Wed, 05 May 2010) | 19 lines
Merged revisions 261274 via svnmerge from
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r261274 | pabelanger | 2010-05-05 12:42:22 -0400 (Wed, 05 May 2010) | 12 lines
Registration fix for SIP realtime.
Make sure realtime fields are not empty.
(closes issue #17266)
Reported by: Nick_Lewis
Patches:
chan_sip.c-realtime.patch uploaded by Nick Lewis (license 657)
Tested by: Nick_Lewis, sberney
Review: https://reviewboard.asterisk.org/r/643/
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r260437 | jpeeler | 2010-04-30 17:36:49 -0500 (Fri, 30 Apr 2010) | 18 lines
Merged revisions 260434 via svnmerge from
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r260434 | jpeeler | 2010-04-30 17:22:46 -0500 (Fri, 30 Apr 2010) | 11 lines
Ensure channel state is not incorrectly set in the case of a very early answer.
The needringing bit was being read in dahdi_read after answering thereby
setting the state to ringing from up. This clears needringing upon answering
so that is no longer possible.
(closes issue #17067)
Reported by: tzafrir
Patches:
needringing.diff uploaded by tzafrir (license 46)
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r260231 | rmudgett | 2010-04-29 17:44:14 -0500 (Thu, 29 Apr 2010) | 33 lines
Merged revisions 260195 via svnmerge from
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r260195 | rmudgett | 2010-04-29 17:11:47 -0500 (Thu, 29 Apr 2010) | 26 lines
DTMF CallerID detection problems.
The code handling DTMF CallerID drops digits on long CallerID numbers and
may timeout waiting for the first ring with shorter numbers.
The DTMF emulation mode was not turned off when processing DTMF CallerID.
When the emulation code gets behind in processing the DTMF digits it can
skip a digit.
For shorter numbers, the timeout may have been too short. I increased it
from 2 seconds to 4 seconds. Four seconds is a typical time between rings
for many countries.
(closes issue #16460)
Reported by: sum
Patches:
issue16460.patch uploaded by rmudgett (license 664)
issue16460_v1.6.2.patch uploaded by rmudgett (license 664)
Tested by: sum, rmudgett
Review: https://reviewboard.asterisk.org/r/634/
JIRA SWP-562
JIRA AST-334
JIRA SWP-901
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r259957 | mmichelson | 2010-04-28 17:34:15 -0500 (Wed, 28 Apr 2010) | 11 lines
Don't override peer context with domain context.
(closes issue #17040)
Reported by: pprindeville
Patches:
asterisk-1.6-bugid17040.patch uploaded by pprindeville (license 347)
Tested by: pprindeville
Review: https://reviewboard.asterisk.org/r/565/
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r259870 | dvossel | 2010-04-28 16:20:03 -0500 (Wed, 28 Apr 2010) | 39 lines
Merged revisions 259858 via svnmerge from
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r259858 | dvossel | 2010-04-28 16:16:03 -0500 (Wed, 28 Apr 2010) | 33 lines
resolves deadlocks in chan_local
Issue_1.
In the local_hangup() 3 locks must be held at the same time... pvt, pvt->chan,
and pvt->owner. Proper deadlock avoidance is done when the channel to hangup
is the outbound chan_local channel, but when it is not the outbound channel we
have an issue... We attempt to do deadlock avoidance only on the tech pvt, when
both the tech pvt and the pvt->owner are locked coming into that loop. By
never giving up the pvt->owner channel deadlock avoidance is not entirely possible.
This patch resolves that by doing deadlock avoidance on both the pvt->owner and the pvt
when trying to get the pvt->chan lock.
Issue_2.
ast_prod() is used in ast_activate_generator() to queue a frame on the channel
and make the channel's read function get called. This function is used in
ast_activate_generator() while the channel is locked, which mean's the channel
will have a lock both from the generator code and the frame_queue code by the
time it gets to chan_local.c's local_queue_frame code... local_queue_frame
contains some of the same crazy deadlock avoidance that local_hangup requires,
and this recursive lock prevents that deadlock avoidance from happening correctly.
This patch removes ast_prod() from the channel lock so only one lock is held during
the local_queue_frame function.
(closes issue #17185)
Reported by: schmoozecom
Patches:
issue_17185_v1.diff uploaded by dvossel (license 671)
issue_17185_v2.diff uploaded by dvossel (license 671)
Tested by: schmoozecom, GameGamer43
Review: https://reviewboard.asterisk.org/r/631/
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r259538 | rmudgett | 2010-04-27 17:18:09 -0500 (Tue, 27 Apr 2010) | 18 lines
Merged revisions 259531 via svnmerge from
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r259531 | rmudgett | 2010-04-27 16:53:07 -0500 (Tue, 27 Apr 2010) | 11 lines
DAHDI "WARNING" message is confusing and vague
"WARNING[28406]: chan_dahdi.c:6873 ss_thread: CallerID feed failed: Success"
Changed the warning to "Failed to decode CallerID on channel 'name'". The
message before it is likely more specific about why the CallerID decode
failed.
SWP-501
AST-283
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r259307 | rmudgett | 2010-04-27 13:29:33 -0500 (Tue, 27 Apr 2010) | 21 lines
Merged revisions 259270 via svnmerge from
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r259270 | rmudgett | 2010-04-27 13:14:54 -0500 (Tue, 27 Apr 2010) | 14 lines
hidecalleridname parameter in chan_dahdi.conf
Issue #7321 implements a new chan_dahdi configuration option. However, a
change mentioned in the issue was never implemented. This is the change
that will allow the feature to work.
I added a note to chan_dahdi.conf.sample about the feature.
(closes issue #17143)
Reported by: djensen99
Patches:
diff.txt uploaded by djensen99 (license NA) (One line change)
Tested by: djensen99
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r257493 | tilghman | 2010-04-15 15:30:15 -0500 (Thu, 15 Apr 2010) | 20 lines
Merged revisions 257467 via svnmerge from
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r257467 | tilghman | 2010-04-15 15:24:50 -0500 (Thu, 15 Apr 2010) | 13 lines
Don't recreate peer, when responding to a repeated deregistration attempt.
When a reply to a deregistration is lost in transmit, the client retries the
deregistration. Previously, this would cause a realtime/autocreate peer to be
loaded back into memory, after it had already been correctly purged. Instead,
we just want to resend the reply without loading the peer.
(closes issue #16908)
Reported by: kkm
Patches:
20100412__issue16908.diff.txt uploaded by tilghman (license 14)
Tested by: kkm
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r257191 | tilghman | 2010-04-13 14:17:48 -0500 (Tue, 13 Apr 2010) | 10 lines
Also unref the pvt when we delete the provisional keepalive job.
(closes issue #16774)
Reported by: kowalma
Patches:
20100315__issue16774.diff.txt uploaded by tilghman (license 14)
Tested by: falves11, jamicque
Review: https://reviewboard.asterisk.org/r/591/
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Converted if statement to a switch statement for clarity.
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r256265 | rmudgett | 2010-04-05 19:39:44 -0500 (Mon, 05 Apr 2010) | 12 lines
Merged revisions 256225 via svnmerge from
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r256225 | rmudgett | 2010-04-05 19:10:16 -0500 (Mon, 05 Apr 2010) | 5 lines
DAHDI/PRI call to pri_channel_bridge() not protected by PRI lock.
SWP-1231
ABE-2163
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r256015 | russell | 2010-04-02 18:46:45 -0500 (Fri, 02 Apr 2010) | 16 lines
Merged revisions 256014 via svnmerge from
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r256014 | russell | 2010-04-02 18:45:56 -0500 (Fri, 02 Apr 2010) | 9 lines
Resolve a deadlock that occurs due to a pointless call to ast_bridged_channel()
(closes issue #16840)
Reported by: bzing2
Patches:
patch.txt uploaded by bzing2 (license 902)
issue_16840.rev1.diff uploaded by russell (license 2)
Tested by: bzing2, russell
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r255410 | russell | 2010-03-30 15:56:26 -0500 (Tue, 30 Mar 2010) | 9 lines
Merged revisions 255409 via svnmerge from
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r255409 | russell | 2010-03-30 15:56:00 -0500 (Tue, 30 Mar 2010) | 2 lines
Don't kill Asterisk if the H323 listener does not start.
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r254718 | russell | 2010-03-25 15:08:40 -0500 (Thu, 25 Mar 2010) | 2 lines
chan_usbradio depends on alsa.
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Specifically, when using the CHANNEL dialplan function, it was
possible to crash Asterisk by trying to get the rtpdest of a
stream type that is not in use by the channel. This commit
fixes that issue.
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r253536 | russell | 2010-03-20 06:33:30 -0500 (Sat, 20 Mar 2010) | 4 lines
Use SHRT_MAX instead of MAXSHORT.
These changes fix build issues I had with this module on FreeBSD.
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r253537 | russell | 2010-03-20 06:39:39 -0500 (Sat, 20 Mar 2010) | 2 lines
Resolve a compiler warning on FreeBSD.
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r253538 | russell | 2010-03-20 06:43:08 -0500 (Sat, 20 Mar 2010) | 2 lines
Resolve compiler warnings on FreeBSD.
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r253540 | russell | 2010-03-20 07:03:07 -0500 (Sat, 20 Mar 2010) | 2 lines
Resolve more compiler warnings on FreeBSD.
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This re-renames ast_rtp_update_source to ast_rtp_new_source
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r252089 | twilson | 2010-03-12 16:04:51 -0600 (Fri, 12 Mar 2010) | 20 lines
Only change the RTP ssrc when we see that it has changed
This change basically reverts the change reviewed in
https://reviewboard.asterisk.org/r/374/ and instead limits the
updating of the RTP synchronization source to only those times when we
detect that the other side of the conversation has changed the ssrc.
The problem is that SRCUPDATE control frames are sent many times where
we don't want a new ssrc, including whenever Asterisk has to send DTMF
in a normal bridge. This is also not the first time that this mistake
has been made. The initial implementation of the ast_rtp_new_source
function also changed the ssrc--and then it was removed because of
this same issue. Then, we put it back in again to fix a different
issue. This patch attempts to only change the ssrc when we see that
the other side of the conversation has changed the ssrc.
It also renames some functions to make their purpose more clear.
Review: https://reviewboard.asterisk.org/r/540/
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r251987 | rmudgett | 2010-03-12 13:40:16 -0600 (Fri, 12 Mar 2010) | 9 lines
Merged revisions 251986 via svnmerge from
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r251986 | rmudgett | 2010-03-12 13:33:22 -0600 (Fri, 12 Mar 2010) | 1 line
Make chan_dahdi wakeup_sub() prototype not conditional.
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r250481 | jpeeler | 2010-03-03 13:06:06 -0600 (Wed, 03 Mar 2010) | 22 lines
Merged revisions 250480 via svnmerge from
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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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r250395 | dvossel | 2010-03-03 12:03:19 -0600 (Wed, 03 Mar 2010) | 22 lines
Merged revisions 250394 via svnmerge from
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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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r250246 | dvossel | 2010-03-02 18:18:28 -0600 (Tue, 02 Mar 2010) | 2 lines
fixes signed to unsigned int comparision issue for FaxMaxDatagram value.
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r249893 | dvossel | 2010-03-02 13:08:38 -0600 (Tue, 02 Mar 2010) | 11 lines
fixes adaptive jitterbuffer configuration
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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r249538 | jpeeler | 2010-03-01 11:11:31 -0600 (Mon, 01 Mar 2010) | 18 lines
Merged revisions 249536 via svnmerge from
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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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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
(closes issue #16789)
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r249235 | kpfleming | 2010-02-27 09:08:35 -0500 (Sat, 27 Feb 2010) | 9 lines
Merged revisions 249234 via svnmerge from
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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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r249101 | mmichelson | 2010-02-26 11:04:58 -0600 (Fri, 26 Feb 2010) | 14 lines
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r249100 | mmichelson | 2010-02-26 11:04:29 -0600 (Fri, 26 Feb 2010) | 8 lines
For T.38 reINVITEs treat a 606 the same as a 488.
(closes issue #16792)
Reported by: vrban
Patches:
t38_606.patch uploaded by vrban (license 756)
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r248397 | dvossel | 2010-02-23 10:34:39 -0600 (Tue, 23 Feb 2010) | 15 lines
Merged revisions 248396 via svnmerge from
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r248396 | dvossel | 2010-02-23 10:26:05 -0600 (Tue, 23 Feb 2010) | 9 lines
fixes invite with replaces deadlock
(closes issue #16862)
Reported by: pwalker
Patches:
replaces_deadlock_1.4 uploaded by dvossel (license 671)
Tested by: pwalker, dvossel
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(closes issue #16470)
Reported by: kjotte
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r228798 | tilghman | 2009-11-09 01:37:52 -0600 (Mon, 09 Nov 2009) | 14 lines
Fix various problems detected with Valgrind.
* chan_console accessed pvts after deallocation.
* The module loader did not check usecount on shutdown, which led to chan_iax2
reading a timer that was already unloaded.
(closes issue #16062)
Reported by: alexanderheinz
Patches:
20091109__issue16062.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
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r247914 | rmudgett | 2010-02-19 11:33:33 -0600 (Fri, 19 Feb 2010) | 62 lines
Merged revisions 247910 via svnmerge from
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r247910 | rmudgett | 2010-02-19 11:18:49 -0600 (Fri, 19 Feb 2010) | 55 lines
Merged revision 247904 from
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r247904 | rmudgett | 2010-02-19 10:49:44 -0600 (Fri, 19 Feb 2010) | 49 lines
Make chan_misdn DTMF processing consistent with other channel technologies.
The processing of DTMF tones on the receiving side of an ISDN channel is
inconsistent with the way it is handled in other channels, especially
DAHDI analog. This causes DTMF tones sent from an ISDN phone to be
doubled at the connected party.
We are using the following 2 options of misdn.conf
1) astdtmf=yes
2) senddtmf=yes
Option one is necessary because the asterisk DSP DTMF detection is better
than mISDN's internal DSP. Not as many false positives.
Option two is necessary to transmit DTMF tones end to end when mISDN
channels are connected to SIP channels with out of band DTMF for example.
The symptom is that DTMF tones sent by an ISDN phone are doubled on the
way through asterisk when two mISDN channels are connected with a Local
channel in between or if it is bridged to an analog channel.
The doubling of DTMF tones is because DTMF is passed inband to asterisk by
the mISDN channel and passed out of band once again after the release of
the DTMF tone. Passing it inband is wrong. Neither an analog channel nor
SIP channel passes DTMF inband if configured to inband DTMF. Analog and
SIP channels filter out the DTMF tones because they use the voice frames
returned by ast_dsp_process. But chan_misdn passes the unfiltered input
voice frames instead.
To overcome one aspect of the problem, the doubling of DTMF tones when two
mISDN channels are directly bridged, someone made an 'optimization', where
in that case the DTMF tone passed out-of-band to the peer channel is not
translated to an inband tone at the transmit side. This optimization is
bad because it does not work in general. For example, analog channels or
mISDN channels when bridged through an intermediary local channel will
generate DTMF tones from out-of-band information. Also, of course, it
must not be done when there is no inband DTMF available.
This patch fixes the issue. Now chan_misdn will filter the received
inband DTMF signal the same as other channel types.
Another change included: No need to build an extra translation path
because ast_process_dsp does it if required.
Patches:
misdn-dtmf.patch
JIRA ABE-2080
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(closes issue #16375)
Reported by: kobaz
(closes issue #16796)
Reported by: kobaz
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r247787 | tilghman | 2010-02-18 15:42:53 -0600 (Thu, 18 Feb 2010) | 17 lines
If the peer record is from realtime, it could be set to 0, due to MySQL not representing NULL well in integer columns.
NULL means the value is not specified for the column, which normally means
the driver uses whatever is the default value. However, on MySQL, placing
a NULL in either a float or integer column results in a retrieval of the 0
value. Hence, users get an errant error on load. This patch suppresses
that error and makes the value as if it was not there.
Note that this cannot be done in the realtime driver, because the lack of
difference between NULL and 0 can only be intepreted correctly by the
driver itself. If we did it in the realtime driver, then it would be
effectively impossible to set any realtime field to 0, because it would act
as if the field were unspecified and possibly take on a different value.
(closes issue #16683)
Reported by: wdoekes
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r246070 | jpeeler | 2010-02-10 10:47:37 -0600 (Wed, 10 Feb 2010) | 22 lines
Change channel state on local channels for busy,answer,ring.
Previously local channels channel state never changed. This became problematic
when the state of the other side of the local channel was lost, for example
during a masquerade. Changing the state of the local channel allows for the
scenario to be detected when the channel state is set to ringing, but the peer
isn't ringing. The specific problem scenario is described in 164201. Although
this was noted on one of the issues, here is the tested dialplan verified to
work:
exten => 9700,1,Dial(Local/*9700@default&Local/0009700@default)
exten => *9700,1,Set(GLOBAL(TESTCHAN)=${CHANNEL:0:${MATH(${LEN(${CHANNEL})}-1):0:2}}1)
exten => *9700,n,wait(3) ;3 works, 1 did not
exten => *9700,n,Dial(SIP/5001)
exten => 0009700,1,Wait(1) ;1 works, 3 did not
exten => 0009700,n,ChannelRedirect(${TESTCHAN},parkedcalls,701,1)
(closes issue #14992)
Reported by: davidw
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r245793 | dvossel | 2010-02-09 17:07:17 -0600 (Tue, 09 Feb 2010) | 18 lines
Merged revisions 245792 via svnmerge from
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r245792 | dvossel | 2010-02-09 16:55:38 -0600 (Tue, 09 Feb 2010) | 12 lines
Fixes iaxs and iaxsl size off by one issue.
2^15 = 32768 which is the maximum allowed iax2 callnumber.
Creating the iaxs and iaxsl array of size 32768 means the maximum
callnumber is actually out of bounds. This causes a nasty crash.
(closes issue #15997)
Reported by: exarv
Patches:
iax_fix.diff uploaded by dvossel (license 671)
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r245578 | tilghman | 2010-02-08 16:31:40 -0600 (Mon, 08 Feb 2010) | 12 lines
Actually use _ASTLDFLAGS in the main/ and channels/ Makefiles.
They were previously passed correctly, but they simply weren't used. This
caused issues with various platforms whose builds needed to pass special
linker flags via the configure script.
(closes issue #16596)
Reported by: pprindeville
Patches:
asterisk-1.6-astldflags.patch uploaded by pprindeville (license 347)
Tested by: tilghman
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r244505 | tilghman | 2010-02-03 12:34:29 -0600 (Wed, 03 Feb 2010) | 8 lines
The chanvar= setting should inherit the entire list of variables, not just the first one.
(closes issue #16359)
Reported by: raarts
Patches:
dahdi-setvars.diff uploaded by raarts (license 937)
Tested by: raarts
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r244443 | dvossel | 2010-02-02 16:27:23 -0600 (Tue, 02 Feb 2010) | 18 lines
fixes crash during T.38 negotiation caused by invalid or missing FaxMaxDatagram field
AST-2010-001
(closes issue #16634)
Reported by: krn
(closes issue #16724)
Reported by: barthpbx
(closes issue #16517)
Reported by: bklang
(closes issue #16485)
Reported by: elsto
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r244071 | tilghman | 2010-02-01 11:53:39 -0600 (Mon, 01 Feb 2010) | 22 lines
Merged revisions 244070 via svnmerge from
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r244070 | tilghman | 2010-02-01 11:46:31 -0600 (Mon, 01 Feb 2010) | 16 lines
Revert previous chan_local fix (r236981) and fix instead by destroying expired frames in the queue.
(closes issue #16525)
Reported by: kobaz
Patches:
20100126__issue16525.diff.txt uploaded by tilghman (license 14)
20100129__issue16525__1.6.0.diff.txt uploaded by tilghman (license 14)
Tested by: kobaz, atis
(closes issue #16581)
Reported by: ZX81
(closes issue #16681)
Reported by: alexr1
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r243780 | russell | 2010-01-28 09:07:23 -0600 (Thu, 28 Jan 2010) | 9 lines
Merged revisions 243779 via svnmerge from
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r243779 | russell | 2010-01-28 09:03:17 -0600 (Thu, 28 Jan 2010) | 2 lines
Fix a bogus third argument to ast_copy_string().
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r243482 | russell | 2010-01-27 11:32:07 -0600 (Wed, 27 Jan 2010) | 13 lines
Fix the ability to specify an OSP token for an outbound IAX2 call.
When this patch was originally submitted, the code allowed for the token to be
set via a channel variable. I decided that a cleaner approach would be to
integrate it into the CHANNEL() function. Unfortunately, that is not a suitable
approach. It's not possible to get the value set on the channel soon enough
using that method. So, go back to the simple channel variable method.
(closes issue #16711)
Reported by: homesick
Patches:
iax-svn.diff uploaded by homesick (license 91)
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1.6.0.
(closes issue #16491)
Reported by: jamicque
Patches:
20100114__issue16491.diff.txt uploaded by tilghman (license 14)
Tested by: jamicque
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r213098 | tilghman | 2009-08-19 16:05:17 -0500 (Wed, 19 Aug 2009) | 9 lines
Better parsing for the "register" line
Allows characters that are otherwise used as delimiters to be used within
certain fields (like the secret).
(closes issue #15008, closes issue #15672)
Reported by: tilghman
Patches:
20090818__issue15008.diff.txt uploaded by tilghman (license 14)
Tested by: lmadsen, tilghman
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r213635 | dvossel | 2009-08-21 16:02:50 -0500 (Fri, 21 Aug 2009) | 5 lines
fixes sip register parsing when user@domain is used
(issue #15008)
(issue #15672)
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r215222 | tilghman | 2009-09-01 16:19:40 -0500 (Tue, 01 Sep 2009) | 3 lines
Fix register such that lines with a transport string, but without an authuser, parse correctly.
(AST-228)
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r215801 | tilghman | 2009-09-02 22:43:51 -0500 (Wed, 02 Sep 2009) | 5 lines
Default the callback extension to "s". This is a regression.
(closes issue #15764)
Reported by: elguero
Change-type: bugfix
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r235132 | dvossel | 2009-12-15 12:43:06 -0600 (Tue, 15 Dec 2009) | 14 lines
reverse minor sip registration regression
A registration regression caused by a code tweak in (issue #14331)
and a bug fix in (issue #15539) caused some sip registration
config entries to be constructed incorrectly. Origially
issue #14331 contained the code tweak as well as a bug fix, but since
the issue was reported as a tweak the bug fix portion was moved into
issue #15539. Both the tweak and the bug fix contained minor incorrect
logic that resulted in some SIP registrations to fail.
(issue #14331)
(issue #15539)
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r242227 | oej | 2010-01-22 10:28:34 +0100 (Fre, 22 Jan 2010) | 11 lines
Merged revisions 242226 via svnmerge from
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r242226 | oej | 2010-01-22 10:19:30 +0100 (Fre, 22 Jan 2010) | 3 lines
Initialize notify_types to NULL
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