[asterisk-commits] lmadsen: tag 1.8.0-beta5 r284546 - /tags/1.8.0-beta5/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Sep 1 14:33:18 CDT 2010
Author: lmadsen
Date: Wed Sep 1 14:33:12 2010
New Revision: 284546
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=284546
Log:
Importing files for 1.8.0-beta5 release.
Added:
tags/1.8.0-beta5/.lastclean (with props)
tags/1.8.0-beta5/.version (with props)
tags/1.8.0-beta5/ChangeLog (with props)
Added: tags/1.8.0-beta5/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.8.0-beta5/.lastclean?view=auto&rev=284546
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--- tags/1.8.0-beta5/ChangeLog (added)
+++ tags/1.8.0-beta5/ChangeLog Wed Sep 1 14:33:12 2010
@@ -1,0 +1,23368 @@
+2010-09-01 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.8.0-beta5 released.
+
+2010-09-01 18:44 +0000 [r284477] Terry Wilson <twilson at digium.com>
+
+ * res/res_srtp.c, res/res_rtp_asterisk.c,
+ include/asterisk/res_srtp.h, main/rtp_engine.c,
+ channels/chan_sip.c: Fix SRTP for changing SSRC and multiple
+ a=crypto SDP lines Adding code to Asterisk that changed the SSRC
+ during bridges and masquerades broke SRTP functionality. Also
+ broken was handling the situation where an incoming INVITE had
+ more than one crypto offer. This patch caches the SRTP policies
+ the we use so that we can change the ssrc and inform libsrtp of
+ the new streams. It also uses the first acceptable a=crypto line
+ from the incoming INVITE. (closes issue #17563) Reported by:
+ Alexcr Patches: srtp.diff uploaded by twilson (license 396)
+ Tested by: twilson Review:
+ https://reviewboard.asterisk.org/r/878/
+
+2010-09-01 18:16 +0000 [r284415-284473] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_config_pgsql.c, /: Merged revisions 284472 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r284472 | tilghman | 2010-09-01 13:13:35 -0500 (Wed, 01
+ Sep 2010) | 5 lines Don't warn on floats and timestamps (closes
+ issue #17082) Reported by: coolmig ........
+
+ * /, channels/chan_sip.c: Merged revisions 284399 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r284399 | tilghman | 2010-08-31 15:18:32 -0500
+ (Tue, 31 Aug 2010) | 14 lines Merged revisions 284393 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r284393 | tilghman | 2010-08-31 15:13:21 -0500 (Tue, 31 Aug 2010)
+ | 7 lines Don't send a devstate change on poke_noanswer if the
+ state did not change. (closes issue #17741) Reported by: schmidts
+ Patches: chan_sip.c.patch uploaded by schmidts (license 1077)
+ ........ ................
+
+2010-08-31 19:00 +0000 [r284318] Leif Madsen <lmadsen at digium.com>
+
+ * configs/say.conf.sample, /: Merged revisions 284317 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r284317 | lmadsen | 2010-08-31 13:59:31 -0500
+ (Tue, 31 Aug 2010) | 15 lines Merged revisions 284316 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r284316 | lmadsen | 2010-08-31 13:57:59 -0500 (Tue, 31 Aug 2010)
+ | 7 lines Update say.conf.sample to match the rules in say.c
+ (closes issue #17835) Reported by: RoadKill Patches:
+ say.conf.sample.patch.rules uploaded by RoadKill (license 933)
+ Tested by: RoadKill ........ ................
+
+2010-08-30 22:28 +0000 [r284281] Tilghman Lesher <tlesher at digium.com>
+
+ * /, apps/app_festival.c: Merged revisions 284280 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r284280 | tilghman | 2010-08-30 17:27:06 -0500 (Mon, 30 Aug 2010)
+ | 11 lines Fix 3 coding errors: 1) After we close FD, we should
+ not be trying to write to it. 2) Call _exit(0), not exit(0), to
+ avoid running shutdown routines in a child. 3) Use endian, not
+ processor, detection to ensure bytes are written in the correct
+ order. (closes issue #15706) Reported by: modelnine Patches:
+ asterisk-1.6.1.1-festival-debug.patch uploaded by modelnine
+ (license 865) Tested by: gmartinez ........
+
+2010-08-29 07:05 +0000 [r284096-284158] Tilghman Lesher <tlesher at digium.com>
+
+ * configs/res_curl.conf.sample (added): Missed adding this file
+
+ * sounds: Also ignore the checksums
+
+ * configs/cel_odbc.conf.sample (added), cel/cel_adaptive_odbc.c
+ (removed), cel/cel_odbc.c (added),
+ configs/cel_adaptive_odbc.conf.sample (removed): Rename CEL
+ adaptive driver to plain driver, since there isn't another ODBC
+ driver (and the other CEL drivers have adaptive capabilities,
+ anyway).
+
+2010-08-28 21:29 +0000 [r284065] Russell Bryant <russell at digium.com>
+
+ * main/manager.c: Be more flexible with whitespace on AMI action
+ headers. Previously, this code required exactly one space to be
+ after the ':' in headers for an AMI action. This now makes
+ whitespace optional, and allows whitespace that is there to vary
+ in amount. (closes issue #17862) Reported by: cmoye Patches:
+ manager.c.patch_trunk uploaded by cmoye (license 858)
+ manager.c.patch_1.8 uploaded by cmoye (license 858) Tested by:
+ cmoye
+
+2010-08-27 22:37 +0000 [r284032] David Vossel <dvossel at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 284002 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r284002 | dvossel | 2010-08-27 17:27:50 -0500
+ (Fri, 27 Aug 2010) | 14 lines Merged revisions 283960 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r283960 | dvossel | 2010-08-27 17:17:26 -0500 (Fri, 27 Aug 2010)
+ | 8 lines Parse all "Accept" headers for SIP SUBSCRIBE requests.
+ (closes issue #17758) Reported by: ibc Patches:
+ multiple_accept_headers_1.4.diff uploaded by dvossel (license
+ 671) ........ ................
+
+2010-08-27 21:33 +0000 [r283951] Russell Bryant <russell at digium.com>
+
+ * pbx/pbx_realtime.c: Print exten at context:priority in verbose
+ messages from pbx_realtime.
+
+2010-08-27 20:31 +0000 [r283882] Jason Parker <jparker at digium.com>
+
+ * main/config.c, addons/res_config_mysql.c, res/res_config_odbc.c,
+ /: Merged revisions 283881 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r283881 | qwell | 2010-08-27 15:30:27 -0500
+ (Fri, 27 Aug 2010) | 15 lines Merged revisions 283880 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r283880 | qwell | 2010-08-27 15:29:11 -0500 (Fri, 27 Aug 2010) |
+ 8 lines Fix issue with decoding ^-escaped characters in realtime.
+ (closes issue #17790) Reported by: denzs Patches:
+ 17790-chunky.diff uploaded by qwell (license 4) Tested by: qwell,
+ denzs ........ ................
+
+2010-08-26 23:47 +0000 [r283770] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_musiconhold.c: Convert MOH to use generic timers. (closes
+ issue #17726) Reported by: lmadsen Patches:
+ 20100825__issue17726__2.diff.txt uploaded by tilghman (license
+ 14) Tested by: tilghman
+
+2010-08-26 15:26 +0000 [r283692] David Vossel <dvossel at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 283691 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r283691 | dvossel | 2010-08-26 10:24:40 -0500
+ (Thu, 26 Aug 2010) | 25 lines Merged revisions 283690 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r283690 | dvossel | 2010-08-26 10:22:28 -0500 (Thu, 26 Aug 2010)
+ | 19 lines Fixed how Asterisk destroys a dialog on channel hangup
+ before invite receives a response. If an ast_channel with a SIP
+ tech pvt hangs up before the sip dialog gets a response to its
+ outgoing INVITE, Asterisk used to pretend_ack the INVITE. This is
+ not rfc compliant and results in confusion at the other endpoint.
+ sip_pretend_ack will ack and remove all the packets in the
+ retransmit queue. This means that the INVITE will stop
+ retransmitting, and that any response to that INVITE that comes
+ after the pretend_ack occurs will be ignored. Instead of faking
+ any sort of acknowledgement for an outgoing INVITE during an
+ internal hangup, we should let the protocol stack process the
+ INVITE transaction and terminate the dialog properly. This is
+ achieved by setting the PENDING_BYE flag. When this flag is used,
+ once the dialog proceeds to an escapable state the transaction
+ will either be canceled with a SIP_CANCEL or completed followed
+ immediately by a BYE. Attempting to do this any other way is
+ incorrect. If the endpoint is not responding to the INVITE
+ request, the INVITE must continue to be retransmitted until it
+ times out which will result in the dialog being destroyed.
+ ........ ................
+
+2010-08-26 13:26 +0000 [r283627-283659] Russell Bryant <russell at digium.com>
+
+ * res/res_odbc.c: Slight improvement to a debug message.
+
+ * keys/iaxtel.pub (removed), keys/freeworlddialup.pub (removed),
+ Makefile: Remove public keys that are no longer useful.
+
+ * configs/manager.conf.sample: Move httptimeout out from in between
+ port and bindaddr.
+
+2010-08-25 22:57 +0000 [r283595] David Vossel <dvossel at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 283594 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r283594 | dvossel | 2010-08-25 17:56:42 -0500 (Wed, 25 Aug 2010)
+ | 7 lines Add to and from tags to NOTIFY dialog-info xml body so
+ pickup can occur. When pedantic mode is used, the dialog-info xml
+ generated during a ringing event must contain the to and from tag
+ values. Otherwise if a pickup occurs using INVITE with replaces,
+ Astrisk will not be able to locate the subscription. ........
+
+2010-08-25 16:12 +0000 [r283561] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_odbc.c: Initialize connect timeout on each time through
+ the loop. (closes issue #17911) Reported by: wurstsalat
+
+2010-08-25 15:54 +0000 [r283559] David Vossel <dvossel at digium.com>
+
+ * channels/sip/include/sip.h, /, channels/chan_sip.c: Merged
+ revisions 283558 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r283558 | dvossel | 2010-08-25 10:52:54 -0500 (Wed, 25 Aug 2010)
+ | 10 lines Asterisk will not advertise session timers are
+ supported when 'session-timers=refuse' is used. Asterisk now
+ dynamically builds the "Supported" header depending on what is
+ enabled/disabled in sip.conf. Session timers used to always be
+ advertised as being supported even when they were disabled in the
+ configuration. This caused problems with some end points. (issue
+ #17005) ........
+
+2010-08-25 14:55 +0000 [r283527] Russell Bryant <russell at digium.com>
+
+ * channels/chan_sip.c: Convert ast_log(LOG_DEBUG, ...) to
+ ast_debug(...)
+
+2010-08-24 20:34 +0000 [r283493] David Vossel <dvossel at digium.com>
+
+ * UPGRADE.txt, configs/sip.conf.sample, channels/sip/include/sip.h:
+ Changes the default behavior for sip.conf's pedantic option from
+ "no" to "yes".
+
+2010-08-24 18:56 +0000 [r283457] Leif Madsen <lmadsen at digium.com>
+
+ * res/res_rtp_asterisk.c, channels/chan_sip.c: Fix issue where TOS
+ is no longer set on RTP packets. Fix issue where the tos is no
+ longer being set on RTP packets through res_rtp_asterisk. (closes
+ issue #17890) Reported by: elguero Patches: qos_18.diff uploaded
+ by elguero (license 37) Review:
+ https://reviewboard.asterisk.org/r/868
+
+2010-08-24 16:11 +0000 [r283382] David Vossel <dvossel at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 283381 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r283381 | dvossel | 2010-08-24 11:07:37 -0500
+ (Tue, 24 Aug 2010) | 18 lines Merged revisions 283380 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r283380 | dvossel | 2010-08-24 11:01:51 -0500 (Tue, 24 Aug 2010)
+ | 11 lines This fix makes sure the ast_channel hangs up correctly
+ when the dialog's PENDING_BYE flag is set. When the pending bye
+ flag is used, it is possible that the dialog will terminate and
+ leave the sip_pvt->owner channel up. This is because we never
+ hangup the ast_channel after sending the SIP_BYE request. When we
+ receive the response for the SIP_BYE we set need_destroy which we
+ would expect to destroy the dialog on the next do_monitor loop,
+ but this is not the case. The dialog will only be destroyed once
+ the owner is hungup even with the need_destroy flag set. This
+ patch sets the softhangup flag on the ast_channel when a SIP_BYE
+ request is sent as a result of the pending bye flag. ........
+ ................
+
+2010-08-24 12:49 +0000 [r283350] Russell Bryant <russell at digium.com>
+
+ * funcs/func_odbc.c: Don't attempt to release a NULL ODBC handle.
+
+2010-08-23 21:33 +0000 [r283319] Tilghman Lesher <tlesher at digium.com>
+
+ * cdr/cdr_adaptive_odbc.c, cdr/cdr_odbc.c, cel/cel_adaptive_odbc.c,
+ /: Merged revisions 283318 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r283318 | tilghman | 2010-08-23 16:32:14 -0500 (Mon, 23 Aug 2010)
+ | 2 lines CDR drivers depend upon res_odbc, not directly on the
+ ODBC libraries ........
+
+2010-08-23 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.8.0-beta4 Released.
+
+2010-08-23 13:35 +0000 [r283177-283241] Russell Bryant <russell at digium.com>
+
+ * configs/cel.conf.sample: Add sample configuration for cel_radius.
+
+ * main/cel.c, include/asterisk/cel.h: Make the AST_CEL_AMA enum
+ match up with the AST_CDR_ ama flag values. Really, having 2
+ enums for this is silly and error prone, demonstrated by the
+ crash that I hit because there was an assumption in the code that
+ the values in each matched up. However, this is a quick fix to
+ get them to match up so it will work.
+
+ * main/cel.c: Don't blow up on an invalid AMA flag.
+
+ * configs/cel_custom.conf.sample: Tack on ${eventextra} to the
+ sample cel_custom.conf.
+
+ * configs/cel_custom.conf.sample: Cut down on excessive quotation.
+
+2010-08-23 12:06 +0000 [r283175] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_stun_monitor.c: Don't fail to start if the config file is
+ missing.
+
+2010-08-23 11:58 +0000 [r283173] Russell Bryant <russell at digium.com>
+
+ * configs/cel_custom.conf.sample: Expand cel_custom.conf.sample.
+ Include the usage of CSV_QUOTE() to ensure data has valid CSV
+ formatting. Also list the special CEL variables that are
+ available for use in the mapping.
+
+2010-08-20 16:51 +0000 [r283050-283125] Richard Mudgett <rmudgett at digium.com>
+
+ * /: Recorded merge of revisions 283124 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r283124 | rmudgett | 2010-08-20 11:48:10 -0500
+ (Fri, 20 Aug 2010) | 16 lines Merged revisions 283123 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ................ r283123 | rmudgett | 2010-08-20 11:46:22 -0500
+ (Fri, 20 Aug 2010) | 9 lines Merged revision 278274 from
+ https://origsvn.digium.com/svn/asterisk/trunk .......... r278274
+ | rmudgett | 2010-07-20 17:38:13 -0500 (Tue, 20 Jul 2010) | 1
+ line Reference correct struct member for unlikely event
+ PRI_EVENT_CONFIG_ERR. .......... ................
+ ................
+
+ * channels/sig_pri.c, /: Merged revisions 283049 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r283049 | rmudgett | 2010-08-20 10:31:03 -0500
+ (Fri, 20 Aug 2010) | 29 lines Merged revisions 283048 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r283048 | rmudgett | 2010-08-20 10:24:36 -0500 (Fri, 20 Aug 2010)
+ | 22 lines Q931 - Sending PROGRESS after sending ALERTING is a
+ protocol error The PRI layer in chan_dadhi will check if a
+ PROGRESS message has already been sent, and not allow sending
+ another (although that is technically allowed by the Q931 spec),
+ however it does not protect against sending an ALERTING and then
+ sending a PROGRESS message, which is a violation of the
+ specification. Most switches don't seem to care too deeply about
+ this, but some do, and will disconnect the call when receiving
+ this invalid sequence. Protocol specification reference:
+ T-REC-Q.931-199805-I page 223, "Figure A.5/Q.931 -- Overview
+ protocol control (network side) point-point (sheet 3 of 8)"
+ (closes issue #17874) Reported by: nic_bellamy Patches:
+ asterisk-1.4-r282537_no-progress-after-alerting.patch uploaded by
+ nic bellamy (license 299)
+ asterisk-1.6.2-r282537_no-progress-after-alerting.patch uploaded
+ by nic bellamy (license 299)
+ asterisk-trunk-r282537_no-progress-after-alerting.patch uploaded
+ by nic bellamy (license 299) ........ ................
+
+2010-08-20 12:45 +0000 [r282979-283013] Russell Bryant <russell at digium.com>
+
+ * configs/cel_adaptive_odbc.conf.sample: Fix a typo in a column
+ name.
+
+ * apps/app_celgenuserevent.c: Add an argument missing from the
+ CELGenUserEvent documentation.
+
+2010-08-19 21:07 +0000 [r282891-282895] David Vossel <dvossel at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 282894 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r282894 | dvossel | 2010-08-19 16:05:54 -0500
+ (Thu, 19 Aug 2010) | 18 lines Merged revisions 282893 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r282893 | dvossel | 2010-08-19 16:03:24 -0500 (Thu, 19 Aug 2010)
+ | 11 lines tos_sip option was not being set correctly When
+ tos_sip is used, the tos of the sip socket is only set correctly
+ if the socket binding changes on a reload. If the binding stays
+ the same but the TOS changes, the new tos value would not take
+ into effect. This patch fixes that. (closes issue #17712)
+ Reported by: nickb ........ ................
+
+ * /, channels/chan_sip.c: Merged revisions 282890 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r282890 | dvossel | 2010-08-19 15:31:22 -0500 (Thu, 19 Aug 2010)
+ | 5 lines fixes sip peer memory leaks in the peer_by_ip table
+ (issue #17798) ........
+
+2010-08-19 20:01 +0000 [r282860] Matthew Nicholson <mnicholson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 282859 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r282859 | mnicholson | 2010-08-19 14:44:00 -0500
+ (Thu, 19 Aug 2010) | 23 lines Merged revisions 277944 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r277944 | pabelanger | 2010-07-19 15:56:07 -0500 (Mon, 19 Jul
+ 2010) | 16 lines Regression with T.38 negotiation Prior to
+ 1.4.26.3 T.38 negotiation worked properly, in the case of the
+ reporter. (issue #16852) Reported by: cfc (closes issue #16705)
+ Reported by: mpiazzatnetbug Patches: issue16705_2.diff uploaded
+ by ebroad (license 878) Tested by: vrban, ebroad, c0rnoTa,
+ samdell3 Review: https://reviewboard.asterisk.org/r/754/ ........
+ ................
+
+2010-08-19 14:44 +0000 [r282826] Tilghman Lesher <tlesher at digium.com>
+
+ * main/netsock2.c: Only output debugging if the debug level is on.
+
+2010-08-19 02:18 +0000 [r282740] Terry Wilson <twilson at digium.com>
+
+ * configs/sip.conf.sample, /: Merged revisions 282730 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r282730 | twilson | 2010-08-18 21:14:28 -0500
+ (Wed, 18 Aug 2010) | 9 lines Merged revisions 282729 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r282729 | twilson | 2010-08-18 21:12:55 -0500 (Wed, 18
+ Aug 2010) | 2 lines Add some documentation about codec
+ negotiation to sip.conf ........ ................
+
+2010-08-18 15:28 +0000 [r282671-282672] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_pri.h: Use the correct type for aoce_delayhangup bit
+ field.
+
+ * channels/chan_dahdi.c: Use the correct operator when calculating
+ the PRI span devstate.
+
+2010-08-18 13:10 +0000 [r282639] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: Properly handle 200 and unknown responses
+ conatined in NOTIFY requests received in response to REFER
+ requests. This patch fixes the way asterisk handles NOTIFY
+ requests received in response to REFER requests. These changes to
+ NOTIFY handler were first introduced in r217482. This new change
+ properly handles the 200 response by queueing an
+ AST_TRANSFER_SUCCESS control frame and also prevents that control
+ frame from being queued when provisional and unknown responses
+ are received. (issue #17486) Reported by: davidw Tested by:
+ mnicholson (issue #12713) Reported by: davidw Review:
+ https://reviewboard.asterisk.org/r/860/
+
+2010-08-18 12:30 +0000 [r282638] Russell Bryant <russell at digium.com>
+
+ * channels/chan_multicast_rtp.c: Split _all_ arguments before
+ parsing them. This fixes multicast RTP paging using linksys mode.
+
+2010-08-18 07:49 +0000 [r282608] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/sig_pri.c, /: Merged revisions 282607 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r282607 | tilghman | 2010-08-18 02:43:14 -0500 (Wed, 18 Aug 2010)
+ | 9 lines Don't warn on callerid when completely text, instead of
+ numeric with localdialplan prefixes. (closes issue #16770)
+ Reported by: jamicque Patches: 20100413__issue16770.diff.txt
+ uploaded by tilghman (license 14) 20100811__issue16770.diff.txt
+ uploaded by tilghman (license 14) Tested by: jamicque ........
+
+2010-08-17 21:36 +0000 [r282543-282577] David Vossel <dvossel at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 282576 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r282576 | dvossel | 2010-08-17 16:35:17 -0500 (Tue, 17 Aug 2010)
+ | 9 lines fixes no default transport for temp peer creation in
+ chan_sip (closes issue #17829) Reported by: falves11 Patches:
+ issue_17829.rev1.txt uploaded by russell (license 2)
+ issue_17829.diff uploaded by dvossel (license 671) Tested by:
+ falves11 ........
+
+ * channels/chan_iax2.c: ACCEPT message should respond with the new
+ FORMAT2 ie (closes issue #17804) Reported by: tpanton
+
+ * include/asterisk/unaligned.h: fixes truncated uint64_t value in
+ put_unaligned_uint64_t() function (issue #17804)
+
+2010-08-16 18:01 +0000 [r282470] Leif Madsen <lmadsen at digium.com>
+
+ * doc/tex/asterisk.tex, doc/tex/sounds.tex (added), /: Merged
+ revisions 282469 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r282469 | lmadsen | 2010-08-16 13:00:09 -0500 (Mon, 16 Aug 2010)
+ | 7 lines Add information about creating sounds files using the
+ sounds tools publically available so that others can create their
+ own sounds prompts using the same tools we use to generate sounds
+ releases. This allows people creating their own prompts to sound
+ consistent with the prompts available from the open source
+ project. SWP-595 ........
+
+2010-08-16 17:53 +0000 [r282468] Terry Wilson <twilson at digium.com>
+
+ * main/channel.c, /: Merged revisions 282467 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r282467 | twilson | 2010-08-16 12:32:01 -0500
+ (Mon, 16 Aug 2010) | 23 lines Merged revisions 282430 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r282430 | twilson | 2010-08-16 12:06:37 -0500 (Mon, 16 Aug 2010)
+ | 16 lines Send a SRCCHANGE indication when we masquerade
+ Masquerading a channel means that the src of the audio is
+ potentially changing, so send a SRCCHANGE so that RTP-based media
+ streams can get a new SSRC generated to reflect the change.
+ Original patch by addix (along with lots of testing--thanks!).
+ (closes issue #17007) Reported by: addix Patches:
+ 1001-reset-SSRC-original-channel.diff uploaded by addix (license
+ 1006) srcchange.diff uploaded by twilson (license 396) Tested by:
+ addix, twilson Review: https://reviewboard.asterisk.org/r/862/
+ ........ ................
+
+2010-08-14 04:53 +0000 [r282366] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_iax2.c, include/asterisk/sched.h: Fix our FRACKing
+ issue with chan_iax2 a different way. Review:
+ https://reviewboard.asterisk.org/r/861/
+
+2010-08-13 23:53 +0000 [r282334] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: PRI CCSS may use a stale dial string for
+ the recall dial string. If an outgoing call negotiates a
+ different B channel than initially requested, the saved original
+ dial string was not transferred to the new B channel. CCSS uses
+ that dial string to generate the recall dial string.
+
+2010-08-13 22:23 +0000 [r282236-282302] David Vossel <dvossel at digium.com>
+
+ * UPGRADE.txt, configs/sip.conf.sample, CHANGES,
+ channels/chan_sip.c: remove current STUN support from chan_sip.c
+ This patch removes the current broken/useless stun support from
+ chan_sip. (closes issue #17622) Reported by: philipp2 Review:
+ https://reviewboard.asterisk.org/r/855/
+
+ * CHANGES: res_stun_monitor and corresponding options CHANGES
+ documentation
+
+ * configs/res_stun_monitor.conf.sample (added),
+ configs/sip.conf.sample, channels/chan_iax2.c,
+ configs/iax.conf.sample, channels/chan_sip.c,
+ include/asterisk/event_defs.h, res/res_stun_monitor.c (added):
+ res_stun_monitor for monitoring network changes behind a NAT
+ device Review: https://reviewboard.asterisk.org/r/854
+
+ * /, channels/chan_sip.c: Merged revisions 282235 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r282235 | dvossel | 2010-08-13 13:54:53 -0500 (Fri, 13 Aug 2010)
+ | 16 lines only do magic pickup when notifycid is enabled A new
+ way of doing BLF pickup was introduced into 1.6.2. This feature
+ adds a call-id value into the XML of a SIP_NOTIFY message sent to
+ alert a subscriber that a device is ringing. This option should
+ only be enabled when the new 'notifycid' option is set... but
+ this was not the case. Instead the call-id value was included for
+ every RINGING Notify message, which caused a regression for
+ people who used other methods for call pickup. (closes issue
+ #17633) Reported by: urosh Patches: chan_sip.txt uploaded by
+ urosh (license ) blf_cid_issue.diff uploaded by dvossel (license
+ 671) Tested by: dvossel, urosh, okrief, alecdavis ........
+
+2010-08-13 16:02 +0000 [r282200-282201] Terry Wilson <twilson at digium.com>
+
+ * configure.ac: Whitespace fix :-/
+
+ * configure, configure.ac: Detect when libsrtp cannot be linked in
+ a shared library The libsrtp build system currently does not
+ produce a shared library or a static library compiled with -fPIC,
+ so on 64-bit systems it is possible that we will get a compile
+ error if libsrtp is installed and res_srtp is selected in
+ menuselect. This patch attempts to detect this situation and
+ provide the user with instructions to work around the problem.
+
+2010-08-12 22:51 +0000 [r282131] Jason Parker <jparker at digium.com>
+
+ * pbx/pbx_config.c, /: Merged revisions 282130 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r282130 | qwell | 2010-08-12 17:50:54 -0500
+ (Thu, 12 Aug 2010) | 9 lines Merged revisions 282129 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.4
+ ........ r282129 | qwell | 2010-08-12 17:49:28 -0500 (Thu, 12 Aug
+ 2010) | 1 line Register CLI commands before parsing config, in
+ case there is a config error. ........ ................
+
+2010-08-12 22:06 +0000 [r282098] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/ccss.h, main/ccss.c: Separate call completion
+ config parameter allocation and default initialization. If you
+ ever have a need to reset the call completion config parameters
+ to defaults, now you can. And no Virginia, C++ idioms do not
+ always work in C.
+
+2010-08-12 20:41 +0000 [r282066] Russell Bryant <russell at digium.com>
+
+ * CHANGES, main/cli.c: Add a "core reload" CLI command. Review:
+ https://reviewboard.asterisk.org/r/859/
+
+2010-08-12 20:15 +0000 [r282047] David Vossel <dvossel at digium.com>
+
+ * CHANGES, include/asterisk/translate.h, main/cli.c,
+ main/translate.c: improved translation paths for wideband codecs
+ The problem I'm addressing is that Asterisk's current method of
+ building the least cost translation paths between codecs does not
+ take into account sample rate. For instance, it was possible for
+ siren14 (a 32khz codec), to contain the a translation path to
+ siren7 (a 16khz audio codec) that goes through slin at 8khz. In
+ this case Asterisk takes a 32khz codec, down samples it to 8khz
+ and then up samples it to 16khz which is terrible regardless if
+ it is computationally less expensive. This patch now builds
+ translation paths that give priority to maintaining the best
+ possible sample rate before taking into consideration
+ computational cost. This patch also adds cli commands to expose
+ what translation paths are actually being used. Changes: 1.
+ Translation paths will never contain a step that changes the
+ sample rate unless absolutely necessary. 2. When choosing the
+ best codec to make two channels compatible. Shared codecs with
+ the highest sample rate are given priority. 3. A new cli command
+ to show all translation paths available for a specific codec
+ 'core show translation paths [codec name]' has been added. 4.
+ 'core show translation' which displays the translation matrix now
+ includes the new higher bit audio codecs in the table. 5. 'core
+ show channel [channel name]' now displays the translation paths
+ if translation is used. (closes issue #16841) Reported by:
+ dvossel Review: https://reviewboard.asterisk.org/r/842/
+
+2010-08-12 18:03 +0000 [r281982-282015] Russell Bryant <russell at digium.com>
+
+ * main/pbx.c: Put back pointer value output for ast_debug(), such
+ that it is only removed for verbose output.
+
+ * main/pbx.c: Remove debugging output from verbose messages.
+ Pointer values to internal objects is not terribly useful to
+ users in the verbose messages about adding extensions and
+ contexts.
+
+2010-08-12 03:03 +0000 [r281913] Jeff Peeler <jpeeler at digium.com>
+
+ * main/channel.c, /: Merged revisions 281912 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r281912 | jpeeler | 2010-08-11 22:01:38 -0500
+ (Wed, 11 Aug 2010) | 27 lines Merged revisions 281911 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r281911 | jpeeler | 2010-08-11 22:00:14 -0500 (Wed, 11 Aug 2010)
+ | 20 lines Ensure SSRC is changed when media source is changed to
+ resolve audio delay. This change causes the SSRC to change right
+ before the channels are bridged, which is what used to happen. It
+ seems that fixes were made to attempt limiting SSRC changes,
+ targeted mainly at sending DTMF. DTMF is not affecting the SSRC
+ with this change. There are two other control frames sent in
+ ast_channel_bridge that probably should also be changed to
+ AST_CONTROL_SRCCHANGE as well, but I'm going to leave this change
+ up to the discretion of resolving issue #17007. For reference -
+ old review implementing new control frame SRCCHANGE:
+ https://reviewboard.asterisk.org/r/540 (closes issue #17404)
+ Reported by: sdolloff Patches: bug17404.patch uploaded by jpeeler
+ (license 325) Tested by: sdolloff ........ ................
+
+2010-08-11 21:12 +0000 [r281875] Leif Madsen <lmadsen at digium.com>
+
+ * configs/say.conf.sample, /: Merged revisions 281873 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r281873 | lmadsen | 2010-08-11 16:09:47 -0500
+ (Wed, 11 Aug 2010) | 14 lines Merged revisions 281819 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r281819 | lmadsen | 2010-08-11 13:28:10 -0500 (Wed, 11 Aug 2010)
+ | 6 lines Add Danish support to say.conf.sample (closes issue
+ #17836) Reported by: RoadKill Patches: say.conf.sample.patch.dk
+ uploaded by RoadKill (license 933) ........ ................
+
+2010-08-11 21:11 +0000 [r281874] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: handle all possible responses to REFER
+ requests (closes issue #17486) Reported by: davidw Patches:
+ Issue17486-counterbid.diff.txt uploaded by davidw (license 780)
+ Tested by: davidw Review: https://reviewboard.asterisk.org/r/837/
+
+2010-08-11 20:30 +0000 [r281870] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_analog.c, channels/sig_analog.h: Fix a call to
+ analog_set_pulsedial() not setting 0 or 1 only. * Also a couple
+ minor tweaks.
+
+2010-08-11 17:54 +0000 [r281764] Leif Madsen <lmadsen at digium.com>
+
+ * configs/say.conf.sample, /: Merged revisions 281763 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r281763 | lmadsen | 2010-08-11 12:54:09 -0500
+ (Wed, 11 Aug 2010) | 14 lines Merged revisions 281762 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r281762 | lmadsen | 2010-08-11 12:51:40 -0500 (Wed, 11 Aug 2010)
+ | 6 lines Allow say.conf to handle large numbers ending with
+ multiple zeros. (closes issue #17833) Reported by: RoadKill
+ Patches: say.conf.sample.patch.largenumbers uploaded by RoadKill
+ (license 933) ........ ................
+
+2010-08-11 17:27 +0000 [r281760] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: Avoid a deadlock in
+ add_header_max_forwards(). Related to r276951
+
+2010-08-11 15:18 +0000 [r281723] Tilghman Lesher <tlesher at digium.com>
+
+ * /, apps/app_readexten.c: Merged revisions 281722 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ........ r281722 | tilghman | 2010-08-11 10:17:20 -0500 (Wed, 11
+ Aug 2010) | 7 lines Only set status TIMEOUT, if we have no
+ digits. (closes issue #15188) Reported by: jcovert Patches:
+ app_readexten.c.patch-1.6.2.8-rc1 uploaded by jcovert (license
+ 551) ........
+
+2010-08-11 13:30 +0000 [r281687] <simon.perreault at viagenie.ca>
+
+ * include/asterisk/netsock2.h, configs/sip.conf.sample,
+ channels/sip/config_parser.c, main/netsock2.c: Fix parsing of
+ IPv6 address literals in outboundproxy (closes issue #17757)
+ Reported by: oej Patches: 17757.diff uploaded by sperreault
+ (license 252) sip.conf.diff uploaded by sperreault (license 252)
+ Tested by: oej
+
+2010-08-10 21:47 +0000 [r281568-281650] Russell Bryant <russell at digium.com>
+
+ * UPGRADE.txt, configs/sip.conf.sample, channels/sip/include/sip.h:
+ Change the default value for alwaysauthreject in sip.conf to
+ "yes". (closes issue #17756) Reported by: oej
+
+ * main/sched.c, /: Merged revisions 281574 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+ r281574 | russell | 2010-08-10 13:04:32 -0500 (Tue, 10 Aug 2010)
+ | 9 lines Don't move the time threshold for running scheduled
+ events on every iteration. Instead, only calculate the time
+ threshold each time ast_sched_runq() is called. (closes issue
+ #17742) Reported by: schmidts Patches: sched.c.patch uploaded by
+ schmidts (license 1077) ........
+
+ * apps/app_dial.c, /: Merged revisions 281567 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r281567 | russell | 2010-08-10 12:47:13 -0500
+ (Tue, 10 Aug 2010) | 15 lines Merged revisions 281566 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r281566 | russell | 2010-08-10 12:45:45 -0500 (Tue, 10 Aug 2010)
+ | 8 lines Reset visible indication after answer. (closes issue
+ #17641) Reported by: klaus3000 Patches:
+ ast1.6.2.9-app_dial-visible_indication.patch.txt uploaded by
+ klaus3000 (license 65) Tested by: schmidts ........
+ ................
+
+2010-08-10 Leif Madsen <lmadsen at digium.com>
+
+ * Asterisk 1.8.0-beta3 Released.
+
+2010-08-10 17:48 +0000 [r281529-281568] Russell Bryant <russell at digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 281567 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.6.2
+ ................ r281567 | russell | 2010-08-10 12:47:13 -0500
+ (Tue, 10 Aug 2010) | 15 lines Merged revisions 281566 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+ r281566 | russell | 2010-08-10 12:45:45 -0500 (Tue, 10 Aug 2010)
+ | 8 lines Reset visible indication after answer. (closes issue
+ #17641) Reported by: klaus3000 Patches:
+ ast1.6.2.9-app_dial-visible_indication.patch.txt uploaded by
+ klaus3000 (license 65) Tested by: schmidts ........
+ ................
+
+ * channels/chan_sip.c: Ensure that the proper external address is
+ used for the RTP destination. (closes issue #17044) Reported by:
+ ebroad Tested by: ebroad Review:
+ https://reviewboard.asterisk.org/r/566/
+
+ * main/cli.c: Resolve a problem with channel name tab completion.
+ Hitting tab without typing any part of a channel name resulted in
+ no results. This now results in getting a full list of active
+ channels, just as it did in previous versions of Asterisk.
+ Review: https://reviewboard.asterisk.org/r/818/
+
+2010-08-10 07:26 +0000 [r281497] TransNexus OSP Development <support at transnexus.com>
+
+ * apps/app_osplookup.c: Fixed the issue caused by EXTEN including
+ user parameters.
+
+2010-08-09 23:04 +0000 [r281466] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_local.c: Add some more stuff to copy from 281429.
+
+2010-08-09 20:47 +0000 [r281432] David Vossel <dvossel at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 281430 via svnmerge from
[... 22620 lines stripped ...]
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