Age | Commit message (Collapse) | Author |
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The PRI channels can no longer change the channel name if a different B
channel is selected during call negotiation. To prevent using the channel
name to infer what B channel a call is using and to avoid name collisions,
the channel name format is changed.
The new channel naming for PRI channels is:
DAHDI/ISDN-<span>-<sequence-number>
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221701 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #15854)
Reported by: dvossel
Patches:
sip_port_config_trunk.diff uploaded by dvossel (license 671)
Tested by: dvossel
Review: https://reviewboard.asterisk.org/r/357/
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r221588 | mnicholson | 2009-10-01 10:24:00 -0500 (Thu, 01 Oct 2009) | 2 lines
Use unsigned ints for portinuri flags.
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FALSE.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r221153 | mnick | 2009-09-30 10:37:39 -0500 (Wed, 30 Sep 2009) | 2 lines
check bounds - prevents for buffer overflow
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r221157 | mnick | 2009-09-30 10:41:46 -0500 (Wed, 30 Sep 2009) | 8 lines
added a new dialplan function 'CSV_QUOTE' and changed the cdr_custom.sample.conf
(closes issue #15471)
Reported by: dkerr
Patches:
csv_quote_14.txt uploaded by mnick (license )
Tested by: mnick
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r221303 | mnick | 2009-09-30 14:02:00 -0500 (Wed, 30 Sep 2009) | 2 lines
changed the prototype definition of csv_quote
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Thanks, Josh.
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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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r221200 | tilghman | 2009-09-30 11:55:21 -0500 (Wed, 30 Sep 2009) | 7 lines
Avoid a potential NULL dereference.
(closes issue #15865)
Reported by: kobaz
Patches:
20090915__issue15865.diff.txt uploaded by tilghman (license 14)
Tested by: kobaz
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name.
(closes issue #14740)
Reported by: pj
Patches:
issue14740_09022009.diff uploaded by seanbright (license 71)
Tested by: seanbright, lmadsen
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We require box numbers, not names as the documentation implies.
(issue #14740)
Reported by: pj
Patches:
__20090729-app_voicemail-documentation.patch uploaded by lmadsen (license 10)
Tested by: seanbright, lmadsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221085 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #14859)
Reported by: atis
Patches:
20090821__issue14859.diff.txt uploaded by tilghman (license 14)
20090925__issue14859__1.6.1.diff.txt uploaded by tilghman (license 14)
Tested by: atis, tilghman
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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ast_cel_report_event would geet a reference to the
bridged channel. However, certain return paths, such
as if CEL was not enabled, would result in a reference
leak. All return paths now properly unref the channel.
(closes issue #15991)
Reported by: mmichelson
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r220907 | mnicholson | 2009-09-29 15:14:29 -0500 (Tue, 29 Sep 2009) | 10 lines
Avoid a deadlock in chanspy, just in case the spyee is masqueraded and chanspy_ds_chan_fixup() is called with the channel locked.
(closes issue #15965)
Reported by: atis
Patches:
chanspy-deadlock-fix1.diff uploaded by mnicholson (license 96)
Tested by: atis
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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fact that app_confbridge does not automatically answer the channel.
(closes issue #15964)
Reported by: shrift
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220904 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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r220717 | seanbright | 2009-09-28 15:09:25 -0400 (Mon, 28 Sep 2009) | 3 lines
When selecting DONT_OPTIMIZE in menuselect, explicitly pass -O0 to the compiler
so we override any default optimization levels for a particular install.
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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/trunk@220718 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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* Removed unneeded and uninitialized service_lock.
* Fixed potential locking imbalance in pri_dchannel():PRI_EVENT_RESTART.
* Fixed verbose message typo in pri_dchannel():PRI_EVENT_RESTART.
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Pointed out to me on IRC by snuff-home. Thanks
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Including version.h here causes this file to get recompiled after
every commit or update, which is not needed.
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ast_channel_get_full() tries to use OBJ_POINTER to optimize name-based
channel lookups, but this will not work properly when the channel's full
name was not supplied; for name-prefix searches, there is no value in
doing a hash-based lookup, and in fact doing so could result in many
channels being skipped.
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JABBER_RECEIVE (along with JabberSend) makes Asterisk interact with users over
XMPP to process calls.
SendText can be used instead of JabberSend in the context of XMPP based voice
channels (chan_gtalk and chan_jingle).
(closes issue #12569)
Reported by: eech55
Tested by: phsultan, asannucci, lmadsen, jtodd, maxgo
Review: https://reviewboard.asterisk.org/r/88/
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default (starting in 1.6.3).
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(closes issue #15939)
Reported by: dvossel
Review: https://reviewboard.asterisk.org/r/375/
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Most of the functionality here is gained simply by setting the feature flag
on the bridge config. However, the dial limit functionality has been moved from
app_dial to the features code and has been made public so both app_dial and
the bridge app can use it.
(closes issue #13165)
Reported by: tim_ringenbach
Patches:
app_bridge_options_r138998.diff uploaded by tim ringenbach (license 540),
modified by me
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docs in the product.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r220288 | tilghman | 2009-09-24 14:39:41 -0500 (Thu, 24 Sep 2009) | 6 lines
Implicitly sending a progress signal breaks some applications.
Call Progress() in your dialplan if you explicitly want progress to be sent.
(Reverts change 216430, closes issue #15957)
Reported by: Pavel Troller on the Asterisk-Dev mailing list
http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html
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r220213 | seanbright | 2009-09-24 14:18:18 -0400 (Thu, 24 Sep 2009) | 1 line
Resolve parallel build warnings.
Reported by Klaus Darilion on the asterisk-dev mailing list.
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escaped.
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r220099 | seanbright | 2009-09-24 10:41:57 -0400 (Thu, 24 Sep 2009) | 2 lines
Remove the remaining bashisms in the Makefile/mkpkgconfig
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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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(closes issue #15739)
Reported by: DLNoah, jeffg
Patches:
20090914__issue15739.diff.txt uploaded by tilghman (license 14)
20090922__issue15739.diff.txt uploaded by tilghman (license 14)
Tested by: DLNoah, jeffg
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provide.
(closes issue #14818)
Reported by: alea-soluciones
Patches:
chan_mgcp-setvars-svn-trunk-r219899.patch uploaded by alea (license 514)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219952 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #15946)
Reported by: jtodd
Patches:
leif-is-a-wuss.txt uploaded by jtodd (license 870)
Tested by: jparker, tilghman, jtodd, russellb, mmichelson, seanbright, kpfleming, and the rest of the usual suspects
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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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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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r219720 | dvossel | 2009-09-21 11:55:53 -0500 (Mon, 21 Sep 2009) | 3 lines
Reverting merge 219520. This change was not necessary.
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r219653 | tilghman | 2009-09-20 12:52:05 -0500 (Sun, 20 Sep 2009) | 8 lines
Really stop the stream, when ast_closestream() is called.
(closes issue #15129)
Reported by: bmh
Patches:
20090918__issue15129.diff.txt uploaded by tilghman (license 14)
Review:
https://reviewboard.asterisk.org/r/372/
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