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r321685 | lmadsen | 2011-06-03 08:17:50 -0500 (Fri, 03 Jun 2011) | 5 lines
Also document the 'queue-minute' option.
(closes issue #19386)
Reported by: juanmol
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Don't send all messages to 's'. Get the destination from the request URI.
(Found using automated test cases).
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r321547 | rmudgett | 2011-06-01 18:11:55 -0500 (Wed, 01 Jun 2011) | 1 line
CDR comment tweaks.
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Asterisk now has protocol independent support for processing text messages
outside of a call. Messages are routed through the Asterisk dialplan.
SIP MESSAGE and XMPP are currently supported. There are options in sip.conf
and jabber.conf that enable these features.
There is a new application, MessageSend(). There are two new functions,
MESSAGE() and MESSAGE_DATA(). Documentation will be available on
the project wiki, wiki.asterisk.org.
Thanks to Terry Wilson for the assistance with development and to David Vossel
for helping with some additional testing.
Review: https://reviewboard.asterisk.org/r/1042/
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r321537 | bbryant | 2011-06-01 16:10:02 -0400 (Wed, 01 Jun 2011) | 8 lines
This patch fixes an issue with using the wrong voicemail folders with greetings.
(closes issue #17871)
Reported by: edhorton
Patches:
digium_bug_17871_2 uploaded by fhackenberger (license 592)
Tested by: edhorton, fhackenberger
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r321528 | may | 2011-06-01 14:40:19 +0400 (Wed, 01 Jun 2011) | 14 lines
Fix double alerting, add forced alerting before answer
Fix double alerting (it wasn't fixed here by issue #18542)
Add forced alerting before connect (if it wasn't before)
Try to send all packets from outgoing queue rather than one only
Call goes into clearing state when disconnect command is received
(closes issue #19361)
Reported by: vmikhelson
Patches:
issue19361-3.patch uploaded by may213 (license 454)
Tested by: vmikhelson
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r321517 | rmudgett | 2011-05-31 15:54:35 -0500 (Tue, 31 May 2011) | 1 line
Update some comments.
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r321515 | dvossel | 2011-05-31 13:52:54 -0500 (Tue, 31 May 2011) | 12 lines
Chan_local locking cleanup.
This patch removes all of the unnecessary deadlock
avoidance loops that occur in chan_local. It also
resolves an issue with a deadlock triggered by
local channel optimizations.
(issue #18028)
Review: https://reviewboard.asterisk.org/r/1231/
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r321511 | lmadsen | 2011-05-31 12:04:47 -0400 (Tue, 31 May 2011) | 8 lines
Enhance NOTICE message to know who couldn't access the dialplan.
(closes issue #19390)
Reported by: lmadsen
Patches:
__20110531-sip-notice-tweak.txt uploaded by lmadsen (license 10)
Tested by: russell
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r321436 | rmudgett | 2011-05-27 19:27:52 -0500 (Fri, 27 May 2011) | 4 lines
Some hagi launch cleanup.
Inspired by issue 19256. This patch would also fix the crash.
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r321392 | rmudgett | 2011-05-27 18:45:41 -0500 (Fri, 27 May 2011) | 12 lines
Crash when using hagi and no servers are available.
When none of the servers returned by the SRV querey respond, asterisk
crashes. The problem is that if the loop over all the SRV entries
finishes then the srv_context has already been cleaned up.
* Make ast_srv_cleanup() check to see if the context is already cleaned
up.
(closes issue #19256)
Reported by: byronclark
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Also revert -r321331 and -r321332.
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r321337 | rmudgett | 2011-05-27 17:06:43 -0500 (Fri, 27 May 2011) | 7 lines
The app_privacy args have undocumented "options" position, interferes with "context" position.
* Add documention for unused "options" position to match existing code.
(closes issue #19273)
Reported by: mdavenport
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r321335 | lmadsen | 2011-05-27 17:54:54 -0400 (Fri, 27 May 2011) | 7 lines
Fix issue with playback of H.261 video.
(closes issue #19379)
Reported by: neutrino88
Patches:
videoprompt.patch uploaded by neutrino88 (license 297)
(changes by russell)
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r321333 | lmadsen | 2011-05-27 17:40:23 -0400 (Fri, 27 May 2011) | 7 lines
Allow parking lot hints and musicclass to be set.
(closes issue #19378)
Reported by: sboily_proformatique
Patches:
pf_parkinghint_music_fix uploaded by sboily proformatique (license 206)
Tested by: russell
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r321330 | rmudgett | 2011-05-27 16:31:25 -0500 (Fri, 27 May 2011) | 8 lines
The app_privacy args have undocumented "options" position, interferes with "context" position.
* Add documention for unused "options" position to match existing code.
The trunk(v1.10) version will remove the unused options position.
(closes issue #19273)
Reported by: mdavenport
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r321273 | jrose | 2011-05-27 09:59:34 -0500 (Fri, 27 May 2011) | 3 lines
markm committed a patch I was working on yesterday, this fixes it to mesh up with suggestions by mnicholson.
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r321211 | alecdavis | 2011-05-27 20:31:15 +1200 (Fri, 27 May 2011) | 16 lines
Fix *8 directed pickup locks system during pickupsound play out
move playout from sip_pickup_thread to bridge using BRIDGE_PLAY_SOUND method,
This stop the clash of 2 threads trying to write audio to same channel.
In addition fixes choppy audio beep in issue 19177.
(issue #18654)
(issue #19177)
Reported by: Docent
Patches:
review1232-1.8.diff.txt alecdavis (license 585)
Tested by: alecdavis
Review: https://reviewboard.asterisk.org/r/1232/
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r321155 | markm | 2011-05-26 17:48:45 -0400 (Thu, 26 May 2011) | 10 lines
Fixed build problem with dev mode enabled, which was caused by commit 321100. Reformulated patch to be more generic.
Moved the sip uri parse variable initalization to parse_uri_full in reqresp_parser.c. This will ensure that any use of parse uri will have null output variables if the parse fails.
(closes issue #19346)
Reported by: kobaz
Tested by: kobaz,JonathanRose
Review: [full review board URL with trailing slash]
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r321100 | markm | 2011-05-26 16:09:35 -0400 (Thu, 26 May 2011) | 11 lines
ast_sockaddr_resolve() in netsock2.c may deref a null pointer
Added a null check in netsock2 ast_sockaddr_resolve() as well as added default initalizers in chan_sip parse_uri_legacy_check() to make sure that invalid uris will make null (and not undefined) user,pass,domain,transport variables
(closes issue #19346)
Reported by: kobaz
Patches:
netsock2.patch uploaded by kobaz (license 834)
Tested by: kobaz, Marquis
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r321044 | rmudgett | 2011-05-26 13:10:17 -0500 (Thu, 26 May 2011) | 1 line
Update ast_sockaddr comment with an important note.
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r321042 | twilson | 2011-05-26 10:29:54 -0700 (Thu, 26 May 2011) | 6 lines
Initialize stack-allocated ast_sockaddrs before use
It is important to always initialize ast_sockaddrs before use--even if they
are passed to ast_sockaddr_copy as the underlying storage could be bigger
than what ends up being copied--leaving part of the data unitialized.
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r320947 | russell | 2011-05-26 10:57:13 -0500 (Thu, 26 May 2011) | 2 lines
Remove some variables that were set but unused.
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The ast_string_field_build_va functions were written to take to separate
va_lists to work around FreeBSD 4 not having va_copy defined.
In the end, we don't support anything using gcc < 3 anyway because we use
va_copy all over the place anyway. This patch just simplifies things by
removing the second va_list function arguments in favor of va_copy.
Review: https://reviewboard.asterisk.org/r/1233/
--This line, and those below, will be ignored--
M include/asterisk/stringfields.h
M main/utils.c
M main/channel.c
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r320883 | rmudgett | 2011-05-25 17:25:18 -0500 (Wed, 25 May 2011) | 17 lines
Native SIP CCSS sends bad CC cancel SUBSCRIBE message.
The SUBSCRIBE message used to cancel a CC request has incorrect To/From
SIP headers. They are reversed and the dialog tags are the same when they
should not be. If pedantic mode was disabled, then the cancel would have
succeeded despite the incorrect message.
* The SIP_OUTGOING flag was not set correctly for the dialog and I had to
move some CC subscribe handling code as a result.
* Initialized the dialog subscribed type to CALL_COMPLETION earlier. If a
CC request SUBSCRIBE message comes in and the CC instance is not found,
the 404 response was duplicated.
JIRA AST-568
JIRA SWP-3493
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r320823 | rmudgett | 2011-05-25 12:06:38 -0500 (Wed, 25 May 2011) | 18 lines
The AMI Newstate event contains different information between v1.4 and v1.8.
The addition of connected line support in v1.8 changes the behavior of the
channel caller ID somewhat. The channel caller ID value no longer time
shares with the connected line ID on outgoing call legs. The timing of
some AMI events/responses output the connected line ID as caller ID.
These party ID's are now separate.
* The ConnectedLineNum and ConnectedLineName headers were added to many
AMI events/responses if the CallerIDNum/CallerIDName headers were also
present.
(closes issue #18252)
Reported by: gje
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1227/
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r320796 | rmudgett | 2011-05-25 11:23:11 -0500 (Wed, 25 May 2011) | 17 lines
Give zombies a safe channel driver to use.
Recent crashes from zombie channels suggests that they need a safe home to
goto. When a masquerade happens, the physical part of the zombie channel
is hungup. The hangup normally sets the channel private pointer to NULL.
If someone then blindly does a callback to the channel driver, a crash is
likely because the private pointer is NULL.
The masquerade now sets the channel technology of zombie channels to the
kill channel driver.
Related to the following issues:
(issue #19116)
(issue #19310)
Review: https://reviewboard.asterisk.org/r/1224/
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Allow Setting / Reading the pickupgroup of a channel with func_channel.c
(closes issue #19045)
Reported by: irroot
Review: https://reviewboard.asterisk.org/r/1148/
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r320716 | twilson | 2011-05-24 17:49:10 -0700 (Tue, 24 May 2011) | 4 lines
Cast data as char * before using S_OR
This is required for compiling successfully under dev mode
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r320650 | rmudgett | 2011-05-23 12:53:44 -0500 (Mon, 23 May 2011) | 16 lines
Add ConnectedLineNum/Name headers to output of AMI action Status.
* Add ConnectedLineNum and ConnectedLineName headers to the output of the
AMI action Status. This makes it easier to find out who the channel is
connected to without having to lookup BridgedChannel or when they are
connected to an application (e.g.: VoiceMail) which has no bridged
channel.
* Bridged channels with no CallerID had "" instead of "<unknown>" output,
that might be a bug as "<unknown>" was what older versions used.
(closes issue #18158)
Reported by: gareth
Patches:
svn-292308.diff uploaded by gareth (license 208)
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r320568 | dvossel | 2011-05-23 11:18:33 -0500 (Mon, 23 May 2011) | 14 lines
Merged revisions 320562 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r320562 | dvossel | 2011-05-23 11:15:18 -0500 (Mon, 23 May 2011) | 9 lines
Adds missing part to the ast_tcptls_server_start fails second attempt to bind patch.
(closes issue #19289)
Reported by: wdoekes
Patches:
issue19289_delay_old_address_setting_tcptls_2.patch uploaded by wdoekes (license 717)
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r320573 | tilghman | 2011-05-23 11:19:32 -0500 (Mon, 23 May 2011) | 7 lines
GNU libiconv uses symbol "libiconv_open" instead of "iconv_open".
(closes issue #19344)
Reported by: rohanl
Patches:
iconv-check.patch uploaded by rohanl (license 1284)
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r320560 | kpfleming | 2011-05-23 10:47:14 -0500 (Mon, 23 May 2011) | 4 lines
Don't generate spurious "No: command not found" messages when running the
configure script on a system that has neither gmime-config nor pkg-config.
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r320504 | jrose | 2011-05-23 09:33:20 -0500 (Mon, 23 May 2011) | 10 lines
Fixes segfault occuring in chan_sip.c at __set_address_from_contact
Checks to see if domain contains anything before sending it off to ast_sockaddr_resolve
which is where the segfault was occuring due to null str.
(closes issue #18857)
Reported by: sybasesql
Review: https://reviewboard.asterisk.org/r/1225/
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r320445 | tilghman | 2011-05-22 18:34:57 -0500 (Sun, 22 May 2011) | 15 lines
Merged revisions 320444 via svnmerge from
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r320444 | tilghman | 2011-05-22 18:25:51 -0500 (Sun, 22 May 2011) | 8 lines
Don't crash when the connection fails.
(closes issue #19250)
Reported by: seadweller
Patches:
20110514__issue19250.diff.txt uploaded by tilghman (license 14)
Tested by: seadweller, sum
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r320338 | dvossel | 2011-05-20 16:39:36 -0500 (Fri, 20 May 2011) | 14 lines
Merged revisions 320271 via svnmerge from
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r320271 | dvossel | 2011-05-20 16:24:48 -0500 (Fri, 20 May 2011) | 8 lines
Fixes issue with ast_tcptls_server_start failing on second attempt to bind.
(closes issue #19289)
Reported by: wdoekes
Patches:
issue19289_delay_old_address_setting_tcptls.patch uploaded by wdoekes (license 717)
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r320237 | rmudgett | 2011-05-20 15:49:03 -0500 (Fri, 20 May 2011) | 27 lines
Merged revisions 320236 via svnmerge from
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r320236 | rmudgett | 2011-05-20 15:44:54 -0500 (Fri, 20 May 2011) | 20 lines
Merged revisions 320235 via svnmerge from
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r320235 | rmudgett | 2011-05-20 15:38:22 -0500 (Fri, 20 May 2011) | 13 lines
The meetme CLI command completion leaves conferences mutex locked.
When issuing a meetme kick CLI command and an invalid (non-existent)
conference number is specified, pressing Tab leaves the conferences mutex
locked and, therefore, all conferences deadlock.
Add missing unlock.
(closes issue #19336)
Reported by: zvision
Patches:
app_meetme.diff uploaded by zvision (license 798)
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r320180 | mnicholson | 2011-05-20 13:48:46 -0500 (Fri, 20 May 2011) | 16 lines
This commit modifies the way polling is done on TLS sockets.
Because of the buffering the TLS layer does, polling is unreliable. If poll is
called while there is data waiting to be read in the TLS layer but not at the
network layer, the messaging processing engine will not proceed until something
else writes data to the socket, which may not occur. This change modifies the
logic around TLS sockets to only poll after a failed read on a non-blocking
socket. This way we know that there is no data waiting to be read from the
buffering layer.
(closes issue #19182)
Reported by: st
Patches:
ssl-poll-fix3.diff uploaded by mnicholson (license 96)
Tested by: mnicholson
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r320162 | jrose | 2011-05-20 13:12:21 -0500 (Fri, 20 May 2011) | 15 lines
Fixes an imapfolder related crash
imapfolders being set in the general section of voicemail would cause the inbox folder name to
change. Since sound file names are made based on the names of the folders, this would cause
the audio related to that folder name to change and if Asterisk attempted to play it, the
channel would instantly hang up when the audio file couldn't be found. This patch searches for
the name of the folder first to leave existing behavior in tact and if that fails, it uses
the normal inbox name to get the sound file instead.
(closes issue #16104)
Reported by: blkline
Review: https://reviewboard.asterisk.org/r/1215/
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r320059 | rmudgett | 2011-05-20 12:03:49 -0500 (Fri, 20 May 2011) | 1 line
Misc comment cleanup in features.c.
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r320057 | rmudgett | 2011-05-20 11:43:02 -0500 (Fri, 20 May 2011) | 19 lines
Crash while transferring a call during DTMF feature timeout.
When a call is being attended transferred during the time between
AST_FRAME_DTMF_BEGIN and AST_FRAME_DTMF_END, the transferred channel
becomes a zombie (so tech data is not available), making ast_dtmf_stream()
segfault when it tries to send the DTMF digit (at least with SIP
channels).
Patch based on feature-end-zombie.patch uploaded by Irontec (license 1256)
* Check for zombies when ast_channel_bridge() returns.
* Guarantee that the fo parameter value is initialized in
ast_channel_bridge() before any returns.
(closes issue #19116)
Reported by: Irontec
Tested by: rmudgett
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Adds a new STRREPLACe function to func_strings.c that allows users to search and replace
against a variable in the dialplan.
(closes issue #18023)
Reported by: wdoekes
Review: https://reviewboard.asterisk.org/r/1219/
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r320007 | rmudgett | 2011-05-20 11:19:01 -0500 (Fri, 20 May 2011) | 2 lines
Change some variable names to make pickup code easier to understand.
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r319997 | rmudgett | 2011-05-20 10:48:25 -0500 (Fri, 20 May 2011) | 25 lines
Crash when using directed pickup applications.
The directed pickup applications can cause a crash if the pickup was
successful because the dialplan keeps executing.
This patch does the following:
* Completes the channel masquerade on a successful pickup before the
application returns. The channel is now guaranteed a zombie and must not
continue executing the dialplan.
* Changes the return value of the directed pickup applications to return
zero if the pickup failed and nonzero(-1) if the pickup succeeded.
* Made some code optimizations that no longer require re-checking the
pickup channel to see if it is still available to pickup.
(closes issue #19310)
Reported by: remiq
Patches:
issue19310_v1.8_v2.patch uploaded by rmudgett (license 664)
Tested by: alecdavis, remiq, rmudgett
Review: https://reviewboard.asterisk.org/r/1221/
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r319938 | jrose | 2011-05-20 08:28:24 -0500 (Fri, 20 May 2011) | 12 lines
Adds legacy_useroption_parsing to address interoperability concerns.
With the new option engaged, Asterisk should interpret user fields with useroptions
contained within the userfield of the uri by stripping them out of the original message
whenever a semicolon is encountered in the userfield string.
(closes issue #18344)
Reported by: danimal
Tested by: jrose
Review: https://reviewboard.asterisk.org/r/1223/
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r319866 | jrose | 2011-05-19 13:32:38 -0500 (Thu, 19 May 2011) | 11 lines
Fix Randomize option on Park()
The randomize option was generally not working like it should have at all on Park().
This patch restores intended functionality.
(closes issue #18862)
Reported by: davidw
Tested by: jrose
Review: https://reviewboard.asterisk.org/r/1222/
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r319812 | markm | 2011-05-19 13:59:01 -0400 (Thu, 19 May 2011) | 9 lines
In cel_odbc, an uninitialized RWLIST is attempted to be locked.
Added INIT and DESTROY for the RWLIST odbc_tables
(closes issue #19331)
Reported by: kobaz
Patches:
odbc_cel.patch uploaded by kobaz (license 834)
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r319758 | rmudgett | 2011-05-19 11:50:48 -0500 (Thu, 19 May 2011) | 21 lines
CCSS generic agent with POTS and ISDN phones fail caller busy call-back test.
If the following is true after a CCSS activation:
* The generic agent is for an analog phone or ISDN phone. (Caller party)
* The called party becomes available.
* The caller party is not available.
When the caller party becomes available, the caller is not alerted to the
called party being available. The generic agent still thinks the caller
is busy.
* Fixed the generic agent device state event subscription to look for all
device states that are considered available.
* Encapsulated the device state test for CCSS generic device available in
cc_generic_is_device_available(). Made the generic agent and monitor use
the new function instead of the manually coded inline equivalent.
JIRA AST-559
JIRA SWP-3462
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r319654 | twilson | 2011-05-18 16:15:58 -0700 (Wed, 18 May 2011) | 22 lines
Merged revisions 319653 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r319653 | twilson | 2011-05-18 16:11:57 -0700 (Wed, 18 May 2011) | 15 lines
Merged revisions 319652 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r319652 | twilson | 2011-05-18 16:04:35 -0700 (Wed, 18 May 2011) | 8 lines
Make sure everyone gets an unhold when a transfer succeeds
Some phones, like the Snom phones, send a hold to the transfer target after
before sending the REFER. We need to make sure that we unhold the parties
that are being connected after the masquerade. If Local channels with the /nm
option are used when dialing the parties, hold music would still be playing on
the transfer target, even after being connected with the transferee.
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r319552 | twilson | 2011-05-18 13:22:36 -0700 (Wed, 18 May 2011) | 11 lines
Unbreak the storing of registrations for restart
The fix for issue 18882 broke retrieving non-realtime peers from the ast_db
on restart/reload. This patch tries to unbreak things while leaving the intent
of the original fix intact.
(closes issue #19318)
Reported by: remiq
Patches:
diff.txt uploaded by twilson (license 396)
Tested by: lmadsen, remiq
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