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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
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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
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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
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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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r319529 | twilson | 2011-05-18 13:05:34 -0700 (Wed, 18 May 2011) | 24 lines
Merged revisions 319528 via svnmerge from
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r319528 | twilson | 2011-05-18 13:02:06 -0700 (Wed, 18 May 2011) | 17 lines
Merged revisions 319527 via svnmerge from
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r319527 | twilson | 2011-05-18 12:56:08 -0700 (Wed, 18 May 2011) | 10 lines
Fix app_dial ring groups
Revert part of r315643. We need to remove the datastore here as well.
The code in bridging code will catch anything that app_dial might miss.
(closes issue #19311)
Reported by: mspuhler
Patches:
issue_19311_no_answer.diff uploaded by elguero (license 37)
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r319469 | rmudgett | 2011-05-17 16:57:56 -0500 (Tue, 17 May 2011) | 22 lines
Merged revision 319468 from
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r319468 | rmudgett | 2011-05-17 16:49:31 -0500 (Tue, 17 May 2011) | 15 lines
The mISDN HDLC mode is prevented on dialed channels.
The use of mISDN HDLC mode is prevented if the mISDN dial technology
option 'h1' is used when config option astdtmf=yes.
There is a bug in channels/misdn/isdn_lib.c which prevents the use of HDLC
mode. Instead of setting the channel to HDLC mode it is set to
transparent(no dsp, no hdlc), although hdlc is not "no hdlc". I.e the
logging message is correct, but the if condition is not.
Make check the nodsp and hdlc flags.
JIRA ABE-2787
JIRA SWP-3437
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The vars were either explicitly or implicitly not used.
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The NEC SV8300 rejects the Q931_IE_TIME_DATE for Q.SIG.
Add option to specify if and how much of the current time is put in
Q931_IE_TIME_DATE.
* Send date/time ie never.
* Send date/time ie date only.
* Send date/time ie date and hour.
* Send date/time ie date, hour, and minute.
* Send date/time ie date, hour, minute, and second.
* Send date/time ie default: Libpri will send date and hhmm only when in
NT PTMP mode to support ISDN phones.
(closes issue #19221)
Reported by: kenner
JIRA SWP-3396
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r319367 | lmadsen | 2011-05-17 07:53:50 -0500 (Tue, 17 May 2011) | 10 lines
Don't create [general] voicemail context when using users.conf
Prior to this patch, app_voicemail would create a [general] context when parsing users.conf.
(closes issue #18891)
Reported by: pdugas
Patches:
app_voicemail-ignore-general.patch uploaded by pdugas (license 1222)
app_voicemail-ignore-general-style-guidelines.patch uploaded by seanbright (license 71)
Tested by: pdugas
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r319365 | lmadsen | 2011-05-17 07:39:37 -0500 (Tue, 17 May 2011) | 6 lines
Make Debian init script lsb compliant
(closes issue #18896)
Reported by: manwe
Patches:
debian_init_lsb.patch uploaded by manwe (license 1223)
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Probably haven't been working for a couple of years. May still need
some more love, but they are now working, both as a hint device and
monitoring a hint. Changes centre around the long ago change
to remove the requirement for a device name in a skinny line, and
changes to the transmit_* functions.
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r319261 | jrose | 2011-05-16 16:00:55 -0500 (Mon, 16 May 2011) | 2 lines
Makes busy detection in dsp.c always allow for at least one frame (20ms) of error so that 200ms tone lengths don't get ignored by single frame error lengths.
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r319259 | rmudgett | 2011-05-16 15:33:37 -0500 (Mon, 16 May 2011) | 13 lines
Deadlock between generic CCSS agent and native ISDN CCSS.
Deadlock can occur when the generic CCSS agent is deleting duplicate CC
offers and the native ISDN CC driver is processing an incoming CC message.
The cc_core_instances container lock cannot be held when an agent or
monitor callback is invoked without the possibility of a deadlock.
* Make kill_duplicate_offers() remove the reference in cc_core_instances
outside of the container lock.
JIRA AST-566
JIRA SWP-3469
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r319204 | twilson | 2011-05-16 13:17:43 -0500 (Mon, 16 May 2011) | 11 lines
Merged revisions 319202 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r319202 | twilson | 2011-05-16 11:00:21 -0700 (Mon, 16 May 2011) | 4 lines
Unlink a peer from peers_by_ip when expiring a registration
Review: https://reviewboard.asterisk.org/r/1218/
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r319145 | dvossel | 2011-05-16 10:57:26 -0500 (Mon, 16 May 2011) | 9 lines
Merged revisions 319144 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r319144 | dvossel | 2011-05-16 10:56:16 -0500 (Mon, 16 May 2011) | 2 lines
Fixes issue with peer ref-counting during handle_request_subscribe.
(closes issue #19293)
Reported by: irroot
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r319142 | mnicholson | 2011-05-16 10:53:26 -0500 (Mon, 16 May 2011) | 8 lines
Make sure tcptls_session exists before dereferencing it.
(closes issue #19192)
Reported by: stknob
Patches:
10-tcptls-unreachable-peer-segfault.patch uploaded by Chainsaw (license 723)
Tested by: vois, Chainsaw
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state of the channel reverts to unknown this should be rejected.
this is important for negotiating T.38 gateway see #13405
This patch adds a option T38_REJECTED that behaves as T38_DISABLED except it reports state rejected.
Trivial Change to res_fax to honnor UNAVAILABLE and REJECTED states.
(closes issue #18889)
Reported by: irroot
Tested by: irroot, darkbasic, mnicholson
Review: https://reviewboard.asterisk.org/r/1115
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r319085 | pabelanger | 2011-05-16 10:35:21 -0400 (Mon, 16 May 2011) | 10 lines
Support gmime-2.4
(closes issue #18863)
Reported by: tzafrir
Patches:
gmime-2.4-18.diff uploaded by tzafrir (license 46)
Tested by: tzafrir
Review: https://reviewboard.asterisk.org/r/1213/
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r319083 | dvossel | 2011-05-16 09:26:33 -0500 (Mon, 16 May 2011) | 5 lines
Fixes Big Endian build issue.
(closes issue #19298)
Reported by: tzafrir
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When called, activatesub first cleans up the active sub and then
handles the sub passed. dialandactivatesub first sets sub->exten
and then calls activatesub. Revise handle_offhook to utilise the
callid sent to chan_skinny. Some other minor fixes especially around
d->hookstate (which still needs some more work).
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r318921 | bbryant | 2011-05-13 14:09:34 -0400 (Fri, 13 May 2011) | 8 lines
Fixes a segmentation fault in dynamic hints when a channel technology isn't
loaded for a hint.
(closes issue #18495)
Reported by: bertrand
Tested by: bertrand
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r318919 | bbryant | 2011-05-13 14:04:50 -0400 (Fri, 13 May 2011) | 10 lines
This patch fixes an issue with SRTP which makes HOLD/UNHOLD impossible when too
much time has passed between sending audio.
(closes issue #18206)
Reported by: bernhardsi
Patches:
res_srtp_unhold.patch uploaded by bernhards (license 1138)
Tested by: bernhards, notthematrix
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r318917 | bbryant | 2011-05-13 13:56:04 -0400 (Fri, 13 May 2011) | 11 lines
This patch allows TCP peers into the ast_db where they were previously
restricted.
(closes issue #18882)
Reported by: cmaj
Patches:
patch-chan_sip-1.8.3-rc2-allow-tcp-peer-store-db-and-readonly-rt-backend.diff.txt
uploaded by cmaj (license 830)
Tested by: cmaj
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r318868 | rmudgett | 2011-05-13 11:28:26 -0500 (Fri, 13 May 2011) | 19 lines
CDR's are being written immediately on caller hangup.
CDR's are being written immediately on caller hangup. The dialplan is not
able to modify it in the h exten. The h exten in the initial context is
not run before closing CDR's when the bridge is unlinked if a macro is
active and does not have an h exten.
* Make ast_bridge_call() check for an h exten in the current context and
if a macro is active then the initial context. The first h exten found is
then run before closing the CDR.
(closes issue #18212)
Reported by: leearcher
Patches:
issue18212_v1.8.patch uploaded by rmudgett (license 664)
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1206/
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There were some issues where if a simple switch was cancelled and a
new switch started before the first had timed out where the d->exten
would be used for both subchannels. This was bad leading to possible
invalid extensions if some digits had been entered in the abandoned
simple switch and the second one was completed before the first timed
out, or the second would be cancelled because d->exten would be set to
nothing on the time out of the first.
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r318720 | mnicholson | 2011-05-12 18:35:51 -0500 (Thu, 12 May 2011) | 4 lines
Handle ipv6 addresses in the sent-by Via: field.
This change fixes a regression in via header parsing and ipv6 handling.
(closes issue #18951)
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r318783 | rmudgett | 2011-05-12 20:47:05 -0500 (Thu, 12 May 2011) | 14 lines
PRI early media won't ring.
And another way to pass early media. Don't indicate that there is inband
information present, just assume that the B channel is connected.
* Restore clearing the dialing flag Rx squelch unconditionally when a
PROCEEDING message comes in.
(closes issue #19268)
Reported by: tbsky
Patches:
issue19268_v1.8.patch uploaded by rmudgett (license 664)
Tested by: tbsky
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r318671 | alecdavis | 2011-05-13 10:52:08 +1200 (Fri, 13 May 2011) | 30 lines
Fix directed group pickup feature code *8 with pickupsounds enabled
Since 1.6.2, the new pickupsound and pickupfailsound in features.conf cause many issues.
1). chan_sip:handle_request_invite() shouldn't be playing out the fail/success audio, as it has 'netlock' locked.
2). dialplan applications for directed_pickups shouldn't beep.
3). feature code for directed pickup should beep on success/failure if configured.
Created a sip_pickup() thread to handle the pickup and playout the audio, spawned from handle_request_invite.
Moved app_directed:pickup_do() to features:ast_do_pickup().
Functions below, all now use the new ast_do_pickup()
app_directed_pickup.c:
pickup_by_channel()
pickup_by_exten()
pickup_by_mark()
pickup_by_part()
features.c:
ast_pickup_call()
(closes issue #18654)
Reported by: Docent
Patches:
ast_do_pickup_1.8_trunk.diff.txt uploaded by alecdavis (license 585)
Tested by: lmadsen, francesco_r, amilcar, isis242, alecdavis, irroot, rymkus, loloski, rmudgett
Review: https://reviewboard.asterisk.org/r/1185/
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