Joshua Colp
6548086af0
Don't assume that the pvt structure will still exist after calling schedule_delivery as it may not. (issue #9278 reported by fmachado)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58923 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-15 18:13:21 +00:00
Russell Bryant
4fc7d081d3
Some people like to put "limitonpeer" instead of "limitonpeers" in their
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configuration. While we're at it, support "limitonpeerz" and
"limitonpeerssssss". (inspired by issue #9172 )
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58906 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-14 19:18:08 +00:00
Russell Bryant
53bc57b467
Add a more basic example setup to the examples section
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2007-03-14 17:04:38 +00:00
Russell Bryant
06f55b3120
Merged revisions 58896 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58896 | russell | 2007-03-14 11:38:48 -0500 (Wed, 14 Mar 2007) | 3 lines
Add a note to the security file that the Asterisk CLI and log files may contain
sensitive information, and that people should keep this in mind.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58897 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-14 16:40:22 +00:00
Russell Bryant
78d178173f
By default, don't attempt to do any CallerID handling at all with SLA because
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it is known to not work properly in some situations. However, add an option to
enable it for those that would like to use it anyway.
The short story behind this is that to properly handle CallerID with SLA, we
need the ability to change the CallerID on an existing call, and we are not
ready to handle that.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-14 16:33:01 +00:00
Tilghman Lesher
2fd66654ac
Issue 9162 - pbx_substitute_variables_helper assumes the buffer is initialized
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to all zeroes. This fixes a case where it wasn't.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58880 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-14 01:47:08 +00:00
Russell Bryant
b7f05aeaaa
Ensure that the blinky lights show that the trunk stopped ringing when the
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trunk hangs up before a station has answered it.
(issue #9234 , reported by francesco_r)
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2007-03-13 23:19:51 +00:00
Russell Bryant
d93c20ac9d
fix the reference to the SLA documentation
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2007-03-13 23:11:08 +00:00
Olle Johansson
0eda92609d
Merged revisions 58847 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58847 | oej | 2007-03-13 12:45:52 +0100 (Tue, 13 Mar 2007) | 2 lines
Issue #9229 - No port in request URI on register to non default SIP ports (neelakantan)
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2007-03-13 11:49:35 +00:00
Olle Johansson
f546453573
Don't hangup the call on OK or errors on MESSAGE and INFO
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inside of a dialog (like video update requests).
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2007-03-13 10:03:03 +00:00
Olle Johansson
83bb6718fc
Issue #9251 - Clear From URI from user attributes (tgrman)
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2007-03-13 09:12:16 +00:00
Joshua Colp
8f379dc693
Blocked revisions 58832 via svnmerge
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r58832 | file | 2007-03-12 12:49:49 -0400 (Mon, 12 Mar 2007) | 2 lines
We can't use the assembler version of fetchadd_int under Intel Macs. (issue #9254 reported by darrell budic)
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2007-03-12 16:52:14 +00:00
Christian Richter
5923e48d7f
Merged revisions 57034,57523,57753,58558 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r57034 | crichter | 2007-02-28 17:09:27 +0100 (Mi, 28 Feb 2007) | 1 line
fixed bugs.digium.com bugs: #9157 and bugs.beronet.com bugs: #302 , #303 , #304
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r57523 | crichter | 2007-03-02 19:32:51 +0100 (Fr, 02 Mar 2007) | 1 line
fixed typo
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r57753 | crichter | 2007-03-04 11:39:50 +0100 (So, 04 Mar 2007) | 1 line
fixed another place where the out_cause was hardcoded to 16
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r58558 | crichter | 2007-03-09 15:43:58 +0100 (Fr, 09 Mar 2007) | 1 line
we can free channel 31 as well, since we can occupy it
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2007-03-12 13:08:06 +00:00
Christian Richter
160cfaf1c6
added UU transceiving and corect handling for rdnis
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58825 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-12 12:43:24 +00:00
Joshua Colp
dd1a0aa1dd
Allow RFC2833 compensation to compensate for even stupider implementations by queueing up the end frame at the start, not the actual end. (issue #8963 reported by AndrewZ)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58783 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-12 01:21:12 +00:00
Joshua Colp
fa866efb5c
Add matchexterniplocally setting which only substitutes your externip/externhost setting if it matches the localnet setting. I know of at least two people who need opposite settings, so I made it an option! (issue #8821 reported by kokoskarokoska)
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2007-03-12 00:51:16 +00:00
Russell Bryant
8ae4e264d1
Fix a few more places in chan_iax2 where the ast_frame used for receiving a
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frame was not properly initialized.
- Interpolating a frame when the jitterbuffer is in use
- decrypting a frame when IAX2 encryption is on
- frames in an IAX2 trunk
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2007-03-10 18:11:11 +00:00
Russell Bryant
442c9f222f
Make the compiler happy and initialize a variable.
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2007-03-10 03:58:27 +00:00
Russell Bryant
e836ed5ea3
Merge some updates to the SLA documentation. I plan to keep working on this
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to explain all of the expected behavior with call handling, configuration
details for specific phones, and other things. However, I got tired of doing
it in plain text, so I switched to using LaTeX. I have included the PDF version.
I haven't been able to get a nice looking plain text version out of it yet, but
I'm not terribly concerned since this is supposed to be more of the manual,
while the plain text sample configuration file is the reference.
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2007-03-09 23:59:10 +00:00
Joshua Colp
76f8e0c489
Fix spelling of unavailable in voicemail documentation. (issue #9248 reported by tensai)
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2007-03-09 21:08:19 +00:00
Joshua Colp
b9124234e9
Merged revisions 58579 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58579 | file | 2007-03-09 15:46:43 -0500 (Fri, 09 Mar 2007) | 2 lines
If we are unable to lookup the host in a c line we have to abort, otherwise the previous data is gone and we will (potentially) have no data when all is said and done.
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2007-03-09 20:49:47 +00:00
Russell Bryant
90aa4ab658
Hang up the channel that put the call on hold in the event processing thread to
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avoid a race condition. Also, if the station originated the call that it is
putting on hold, don't hang up the trunk if it was the only station on the call
and it is hanging up due to hold and not a normal hangup.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58512 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-08 22:15:15 +00:00
Russell Bryant
2f54661cf8
Add a missing break statement so that handling the above event does not
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incorrectly destroy the channel. (issue #9242 , andrew)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58510 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-08 22:06:54 +00:00
Tilghman Lesher
ec21f50132
Fix segfault (Issue 9236)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58479 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-08 21:33:03 +00:00
Russell Bryant
ed9b9c6791
Refactor hold handling a bit so that it does not require keeping the call up
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when a call is put on hold.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58474 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-08 20:54:56 +00:00
Joshua Colp
2b3c0dc1e4
Make early SDP seeding even smarter! We have to check codecs in the make_compatible function too. (issue #9221 reported by marcelbarbulescu)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-08 18:01:00 +00:00
Joshua Colp
add0cad065
Merged revisions 58388 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58388 | file | 2007-03-08 11:04:58 -0500 (Thu, 08 Mar 2007) | 2 lines
Only print out debug message if the definition that makes the variables shows up was actually defined. (issue #9233 reported by serginuez)
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2007-03-08 16:07:10 +00:00
Kevin P. Fleming
690da7f074
this change was not needed; fclose() handles closing the file descriptor already
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2007-03-08 13:23:46 +00:00
Kevin P. Fleming
37af77e79f
fix a compiler warning, and overwriting 'res' value
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58352 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-08 13:17:42 +00:00
Kevin P. Fleming
8a2e3d0b8b
fix two cases where HTTP session file descriptors would not be closed
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58351 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-08 13:17:17 +00:00
Russell Bryant
5603ca69eb
If we receive ZT_EVENT_REMOVED, destroy the specified channel.
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(issue #7256 , tzafrir)
Also, update the configure script to make sure that we don't try to build
chan_zap if the installed version of zaptel does not include ZT_EVENT_REMOVED.
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2007-03-08 01:01:46 +00:00
Russell Bryant
f34c30860e
(This bug was reported to me by Kinsey Moore)
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Merged revisions 58242 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58242 | russell | 2007-03-07 12:17:07 -0600 (Wed, 07 Mar 2007) | 7 lines
Fix a problem where the Asterisk channel name could be that of the wrong IAX2
user for a call. This is because the first step of choosing this name is to
look for an IAX2 peer that happens to have the same IP/port number that this
call is coming from and assuming that is it. However, this is not always
correct. So, I have made it change this name after authentication happens
since at that point, we have an exact match.
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2007-03-07 18:19:19 +00:00
Joshua Colp
240355d78b
Ensure we have (or should have) at least one matching codec before attempting early bridge SDP seeding. (issue #9221 reported by marcelbarbulescu)
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2007-03-07 17:52:58 +00:00
Russell Bryant
b393ff7feb
Blocked revisions 58167 via svnmerge
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r58167 | russell | 2007-03-06 18:27:04 -0600 (Tue, 06 Mar 2007) | 2 lines
Fix a misplaced block of code in the 1.2 version of the patch to fix issue #8977
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2007-03-07 00:27:59 +00:00
Russell Bryant
97fe798e7e
Merged revisions 58164 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58164 | russell | 2007-03-06 18:20:13 -0600 (Tue, 06 Mar 2007) | 4 lines
If the channels acquired using the manager Redirect action are not up, then
don't attempt to do anything with them. It could lead to weird behavior,
including crashes. (issue #8977 )
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2007-03-07 00:25:19 +00:00
Steve Murphy
81ef9a4f18
Merged revisions 58115 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58115 | murf | 2007-03-06 15:52:52 -0700 (Tue, 06 Mar 2007) | 1 line
Fix for 9220: Eyebeam cannot renew subscriptions for presence info. Reason: re-SUBSCRIBE requests don't include Accept headers, which the rfc says are optional (to put it tersely), (it uses MAY), and luckily, the sip_pvt struct has the format info stored, so we simply leave it if the format is set, and the accept header null.
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2007-03-06 23:10:14 +00:00
Russell Bryant
dd920562ee
Clarify the documentation of the dialout and sendvoicemail options.
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(issue #9000 , caio1982 and serge-v)
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2007-03-06 23:00:57 +00:00
Olle Johansson
275abf4e08
Merged revisions 58052 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58052 | oej | 2007-03-06 21:33:21 +0100 (Tue, 06 Mar 2007) | 2 lines
Change error message to proper message
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2007-03-06 20:37:07 +00:00
Russell Bryant
905238839b
Return an error of transmit_response is called without a session.
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(issue #9002 )
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2007-03-06 18:01:20 +00:00
Joshua Colp
94c9dcda60
Since chan_iax2 does not support reception of DTMF with duration ensure that it is set to 0 on the frame. (issue #8521 reported by gdhgdh)
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2007-03-05 19:19:07 +00:00
Joshua Colp
a5b62f15c3
Don't create a listen channel and record the conference unless the option is turned on. (issue #9204 reported by francesco_r)
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2007-03-05 18:39:28 +00:00
Joshua Colp
2feaf74cdd
Merged revisions 57869 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r57869 | file | 2007-03-05 12:49:18 -0500 (Mon, 05 Mar 2007) | 2 lines
Make create_dirpath use our standard for return values. -1 is failure, 0 is success. (issue #9205 reported by ballares)
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2007-03-05 17:52:03 +00:00
Steve Murphy
85b9c4f565
Merged revisions 57825 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r57825 | murf | 2007-03-05 07:53:57 -0700 (Mon, 05 Mar 2007) | 1 line
Fixed a typo introduced via 9156 (either the gotos or their doc strings are wrong)
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2007-03-05 15:20:17 +00:00
Joshua Colp
127e1095f1
Don't allow a NULL pointer to reach ast_frdup. (issue #9155 reported by cmaj)
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2007-03-05 04:19:53 +00:00
Joshua Colp
a00f75604b
Don't reference a potentially NULL pointer. (issue #9199 reported by klolik)
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2007-03-05 03:35:03 +00:00
Joshua Colp
44dd29ff61
Preserve marker bit when P2P bridging. (issue #9198 reported by edgreenberg)
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2007-03-05 03:22:17 +00:00
Steve Murphy
b899145141
Updated the regression tests
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2007-03-03 15:31:12 +00:00
Tilghman Lesher
3687771760
Merged revisions 57648 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r57648 | tilghman | 2007-03-03 00:36:55 -0600 (Sat, 03 Mar 2007) | 2 lines
Memory leak of a list, if call recording was abandoned
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2007-03-03 06:45:00 +00:00
Dwayne M. Hubbard
9a47a4712f
submitted patch for Georgian language, issue 9010, submitted by Alexander Shaduri
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2007-03-03 00:59:24 +00:00
Russell Bryant
3b6dc39807
add missing configuration template. Thanks to Lacy Moore on asterisk-users for pointing this out\!
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2007-03-03 00:02:29 +00:00