https://origsvn.digium.com/svn/asterisk/trunk
................
r190663 | russell | 2009-04-27 14:08:12 -0500 (Mon, 27 Apr 2009) | 22 lines
Merged revisions 190661-190662 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r190661 | russell | 2009-04-27 14:00:54 -0500 (Mon, 27 Apr 2009) | 9 lines
Resolve a crash in res_smdi when used with chan_dahdi.
When chan_dahdi goes to get an SMDI message, it provides no search criteria.
It just grabs the next message that arrives. This code was written with the
SMDI dialplan functions in mind, since that is now the preferred method of
using SMDI. However, this broke support of it being used from chan_dahdi.
(closes AST-212)
........
r190662 | russell | 2009-04-27 14:03:59 -0500 (Mon, 27 Apr 2009) | 2 lines
Fix a typo from 190661.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@190665 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r190626 | mmichelson | 2009-04-27 11:37:51 -0500 (Mon, 27 Apr 2009) | 14 lines
Allow for a position to be specified when entering a queue.
This would allow for one to add a caller to a specific place in the
queue instead of just placing the caller in the back every time. To help
facilitate some interesting manipulations, a new channel variable called
QUEUEPOSITION has been added. When a caller is removed from a queue, his
position in that queue is stored in the QUEUEPOSITION variable. One such
strategy an administrator can employ is to allow for the removal of a caller
from one queue followed by the insertion of the same caller into a separate
queue in the same position.
Review: http://reviewboard.digium.com/r/189
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@190628 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r190577 | mmichelson | 2009-04-27 09:46:14 -0500 (Mon, 27 Apr 2009) | 6 lines
Remove nonexistent option from sip.conf.sample.
The option to choose which connected line header to
use is not 'rpid_header' but 'sendrpid'
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@190579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r190287 | file | 2009-04-23 16:15:30 -0300 (Thu, 23 Apr 2009) | 13 lines
Merged revisions 190286 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r190286 | file | 2009-04-23 16:13:18 -0300 (Thu, 23 Apr 2009) | 6 lines
Fix a bug in chan_local glare hangup detection.
If both sides of a Local channel were hung up at around the same time it was
possible for one thread to destroy the local private structure and have the other thread
immediately try to remove the already freed structure from the local channel list.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@190291 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r190057 | jpeeler | 2009-04-22 16:15:55 -0500 (Wed, 22 Apr 2009) | 9 lines
Fix building of chan_h323 with gcc-3.3
There seems to be a bug with old versions of g++ that doesn't allow a structure
member to use the name list. Rename list member to group_list in ast_group_info
and change the few places it is used.
(closes issue #14790)
Reported by: stuarth
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@190063 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r189993 | jpeeler | 2009-04-22 14:23:49 -0500 (Wed, 22 Apr 2009) | 18 lines
Make chan_h323 respect packetization settings
Previously, packetization settings were ignored and now they are not. A new
config option 'autoframing' has been added to mirror the way chan_sip handles
it. Turning on the autoframing option (available both as a global option or per
peer) overrides the local settings with the remote packetization settings.
Testing was performed with varying packetization levels with the following
codecs: ulaw, alaw, gsm, and g729.
(closes issue #12415)
Reported by: pj
Patches:
2009012200_h323packetization.diff.txt uploaded by mvanbaak (license 7),
modified by me
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@189996 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r189911 | tilghman | 2009-04-22 11:01:30 -0500 (Wed, 22 Apr 2009) | 7 lines
Do not continue to receive DTMF, when the channel is hungup and about to be destroyed.
(closes issue #14858)
Reported by: barryf
Patches:
20090421__bug14858.diff.txt uploaded by tilghman (license 14)
Tested by: barryf
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@189913 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r189850 | mvanbaak | 2009-04-22 16:30:47 +0200 (Wed, 22 Apr 2009) | 19 lines
Merged revisions 189849 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r189849 | mvanbaak | 2009-04-22 16:29:28 +0200 (Wed, 22 Apr 2009) | 12 lines
replace sed with tr to remove \r from downloaded file
On some systems, sed does not recognize \r in the pattern the way it
was used here.
Use tr instead because this works the same across systems.
(closes issue #14936)
Reported by: leobrown
Patches:
2009042201_14936.diff.txt uploaded by mvanbaak (license 7)
Tested by: leobrown, mvanbaak
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@189852 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r189771 | dvossel | 2009-04-21 15:28:37 -0500 (Tue, 21 Apr 2009) | 11 lines
Fixes segfault when switching UDP to TCP in sip.conf after reload.
If transport in sip.conf is switched from UDP to TCP, Asterisk segfaults right after issuing a sip reload. The problem is the socket type is changed to TCP but the fd may still be present for UDP. Later, when the TCP session should be created or set using an existing one, it isn't because the old file descriptor is still present. Now every time transport is changed during a sip.conf reload, the file descriptor is set to -1, signifying it must be created or found.
(closes issue #14727)
Reported by: pj
Tested by: dvossel
Review: http://reviewboard.digium.com/r/229/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@189774 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r189629 | dbailey | 2009-04-21 09:28:04 -0500 (Tue, 21 Apr 2009) | 10 lines
Merged revisions 189601 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r189601 | dbailey | 2009-04-21 09:00:55 -0500 (Tue, 21 Apr 2009) | 3 lines
Add check in configure script to check for GLOB_NOMAGIC and GLOB_BRACE in glob.h
This allows config.c to compile when linked against uclibc that does not support these parameters
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@189642 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r189464 | seanbright | 2009-04-20 17:09:59 -0400 (Mon, 20 Apr 2009) | 20 lines
Merged revisions 189462 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r189462 | seanbright | 2009-04-20 16:58:39 -0400 (Mon, 20 Apr 2009) | 13 lines
Properly handle @s within hints in AEL.
AEL was not handling the case of a device hint containing an @ symbol, which
caused parking hints (e.g. hint(park:exten@context)) to error out the parser.
This patch makes AEL treat the @ the same way it treats colon and ampersand
now, meaning the characters are included in verbatim.
(closes issue #14941)
Reported by: bpgoldsb
Patches:
bug14941.patch uploaded by seanbright (license 71)
Tested by: bpgoldsb
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@189531 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r189419 | dbailey | 2009-04-20 14:28:16 -0500 (Mon, 20 Apr 2009) | 11 lines
Merged revisions 189391 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r189391 | dbailey | 2009-04-20 14:10:56 -0500 (Mon, 20 Apr 2009) | 4 lines
Clean up problem with manager implementation of mmap where it was not testing against MAP_FAILED response.
Got rid of shadowed variable used in processign the mmap results.
Change test of mmap results to compare against MAP_FAILED
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@189422 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r189350 | file | 2009-04-20 14:05:15 -0300 (Mon, 20 Apr 2009) | 10 lines
Fix a bug with non-UDP connections that caused dialogs to not get freed.
This issue crept up because of a reference count issue on non-UDP based dialogs.
The dialog reference count was increased when transmitting a packet reliably but never
decreased. This caused the dialog structure to hang around despite being unlinked from
the dialogs container.
(closes issue #14919)
Reported by: vrban
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@189352 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r189204 | dvossel | 2009-04-17 20:28:45 -0500 (Fri, 17 Apr 2009) | 18 lines
Merged revisions 189203 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r189203 | dvossel | 2009-04-17 20:27:19 -0500 (Fri, 17 Apr 2009) | 12 lines
Fixed autologoff in agents.conf not working when agent logs in via AgentLogin app
An agent logs in by calling an extension that calls the AgentLogin app. In agents.conf ackcall=always is set, so when they get a call they have the choice to either acknowledge it or ignore it. autologoff=10 is set as well, so if the agent ignores the call over 10sec one may assume that the agent should be logged out (and in this case hungup on as well), but this was not happening.
(closes issue #14091)
Reported by: evandro
Patches:
autologoff.diff uploaded by dvossel (license 671)
Review: http://reviewboard.digium.com/r/225/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@189206 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r189097 | mmichelson | 2009-04-17 15:20:23 -0500 (Fri, 17 Apr 2009) | 13 lines
Prevent a crash when SIP blonde transferring an unbridged call.
If one attempts to use the attended transfer button on a SIP phone
to transfer an unbridged call (such as a call to an IVR) but hangs
up while the target of the transfer is still ringing, we need to not
crash.
The problem was that ast_hangup was called from outside the channel
thread.
AST-211
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@189103 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r188947 | file | 2009-04-17 11:44:56 -0300 (Fri, 17 Apr 2009) | 22 lines
Merged revisions 188946 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r188946 | file | 2009-04-17 11:41:25 -0300 (Fri, 17 Apr 2009) | 15 lines
Fix a bug where a value used to create the channel name was bogus.
This commit fixes the scenario where an incoming call is authenticated
using a peer entry. Previously the channel name was created using either
the username setting from the sip.conf entry or the IP address that the
call came from. Now the channel name will be created using the peer name
itself. This commit will not change the way the channel name is generated
for users or friends.
(closes issue #14256)
Reported by: Nick_Lewis
Patches:
chan_sip.c-chname.patch uploaded by Nick (license 657)
Tested by: Nick_Lewis, file
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@188949 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r188942 | mmichelson | 2009-04-17 09:33:50 -0500 (Fri, 17 Apr 2009) | 5 lines
Fix a spacing issue that I claimed I would when I committed this code.
Nothing major though.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@188944 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r188901 | mmichelson | 2009-04-17 08:29:33 -0500 (Fri, 17 Apr 2009) | 28 lines
Several fixes to the extenpatternmatchnew logic.
1. Differentiate between literal characters in an extension
and characters that should be treated as a pattern match. Prior to
these fixes, an extension such as NNN would be treated as a pattern,
rather than a literal string of N's.
2. Fixed the logic used when matching an extension with a bracketed
expression, such as 2[5-7]6.
3. Removed all areas of code that were executed when NOT_NOW was
#defined. The code in these areas had the potential to crash, for
one thing, and the actual intent of these blocks seemed counterproductive.
4. Fixed many many coding guidelines problems I encountered while looking
through the corresponding code.
5. Added failure cases and warning messages for when duplicate extensions
are encountered.
6. Miscellaneous fixes to incorrect or redundant statements.
(closes issue #14615)
Reported by: steinwej
Tested by: mmichelson
Review: http://reviewboard.digium.com/r/194/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@188903 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r188647 | dvossel | 2009-04-15 17:10:04 -0500 (Wed, 15 Apr 2009) | 18 lines
Merged revisions 188646 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r188646 | dvossel | 2009-04-15 17:08:40 -0500 (Wed, 15 Apr 2009) | 12 lines
National prefix inserted even when caller ID not available
When the caller ID is restricted, the expected behavior is for the caller id to be blank. In chan_dahdi, the national prefix is placed onto the callers number even if its restricted (empty) causing the caller id to be the national prefix rather than blank.
(closes issue #13207)
Reported by: shawkris
Patches:
national_prefix.diff uploaded by dvossel (license 671)
Review: http://reviewboard.digium.com/r/220/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@188649 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r188544 | mmichelson | 2009-04-15 10:24:50 -0500 (Wed, 15 Apr 2009) | 9 lines
Make the cancellation of the dial timeout on a call forward optional.
This introduces the 'z' option to app_dial. With it set, a call forward
will cancel any timeout originally set for this instance of the Dial
application.
AST-207
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@188547 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r188342 | jpeeler | 2009-04-14 10:54:16 -0500 (Tue, 14 Apr 2009) | 33 lines
Add service maintenance message support
This is the companion commit to libpri r732. Service messages are now supported
for switch types 4ess/5ess. A new option service_message_support has been added
to chan_dahdi.conf and is noted in the sample config file. The service message
support is turned off by default. The current implementation relies on AstDB
to keep track of channel state, which allows the statuses to be preserved
across Asterisk restarts. Below is a description of the storage format.
The state and reason for the service state are in the form <state>:<reason>,
where:
<state> ::= { 'O' } // 'O' – Out Of Service
<reason> ::= { '0' | '1' | '2' | '3' }, where:
'0' – No reason (backwards compatibility)
'1' – NEAR END
'2' – FAR END
'3' – both NEAR and FAR END
The new CLI commands to handle channel service state are:
pri service disable channel <chan>
pri service enable channel <chan>
Many people contributed to the development of this functionality. Because I
entered at the very end I do not know the exact history. Special thanks to
all who moved the bug forward one way or another:
cmaj, PCadach, markster, mattf, drmac, MikeJ, serge-v, murf, kanelbullar, Seb7,
tilghman, lmadsen, and especially dhubbard (he answered lots of my questions
and did a large portion of the work)
(closes issue #3450)
Reported by: cmaj
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@188344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r188247 | file | 2009-04-14 10:14:21 -0300 (Tue, 14 Apr 2009) | 7 lines
Fix a bug with the change I made yesterday to outbound proxy support.
Per discussion with oej on IRC we need the actual IP address, not the
outbound proxy IP address, in the sa field. Upon further inspection
this should make the behaviour of all other uses of the outbound proxy
in the code.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@188254 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r188067 | file | 2009-04-13 13:28:06 -0300 (Mon, 13 Apr 2009) | 10 lines
Fix a bug where using an outbound proxy would cause the local address to be 127.0.0.1.
Copy the outbound proxy IP address into the SIP dialog structure as the IP address we will
be sending to. This has to be done because the logic that determines what local IP address to use
in the SIP messages is not aware of an outbound proxy being in place. It only knows what IP address
we are sending to.
(closes issue #12006)
Reported by: mnicholson
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@188069 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r188032 | mmichelson | 2009-04-13 09:17:56 -0500 (Mon, 13 Apr 2009) | 6 lines
Set all queue variables on both the caller and member channels.
This allows for the variables to be accessed if a member macro is run.
Thanks to Grigoriy Puzankin for bringing this up on the -dev list.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@188038 65c4cc65-6c06-0410-ace0-fbb531ad65f3
