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r168579 | mmichelson | 2009-01-13 16:30:59 -0600 (Tue, 13 Jan 2009) | 13 lines
Clarify a message that app_queue prints and change to a debug-level message
The "No one is answering..." verbose message contained 3 numbers that were not
explained in any way to whoever was viewing the message. It is more helpful now
since the message explains what the numbers mean. Also, the message has been
downgraded to "DEBUG" level.
(closes issue #14172)
Reported by: caio1982
Patches:
queue_answering_debug.diff uploaded by caio1982 (license 22)
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r168575 | mmichelson | 2009-01-13 15:18:13 -0600 (Tue, 13 Jan 2009) | 13 lines
Allow specifying a port number in the user portion of a register => line in sip.conf
With this commit, a register => line in sip.conf may contain a port number in the
"user" section of the line. Please see CHANGES and sip.conf.sample for more
details regarding this.
(closes issue #14198)
Reported by: Nick_Lewis
Patches:
chan_sip.c-domainport2.patch uploaded by Nick (license 657)
Tested by: Nick_Lewis
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r168523 | mmichelson | 2009-01-12 17:12:30 -0600 (Mon, 12 Jan 2009) | 11 lines
bump the verbosity of a message in srv.c up by one. It used to be
at this level prior to a large patch merge which converted ast_verbose
calls to ast_verb
(closes issue #14221)
Reported by: jcovert
Patches:
srv.c.patch uploaded by jcovert (license 551)
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r168508 | jpeeler | 2009-01-12 14:53:04 -0600 (Mon, 12 Jan 2009) | 15 lines
Merged revisions 168507 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168507 | jpeeler | 2009-01-12 14:26:22 -0600 (Mon, 12 Jan 2009) | 9 lines
(closes issue #12269)
Reported by: IgorG
Tested by: denisgalvao
This gits rid of the notion of an owning_app allowing the request and hangup to be initiated by different threads. Originating from an active agent channel requires this. The implementation primarily changes __login_exec to wait on a condition variable rather than a lock.
Review: http://reviewboard.digium.com/r/35/
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r167973 | twilson | 2009-01-08 19:15:43 -0600 (Thu, 08 Jan 2009) | 2 lines
Set ORIGINATE_STATUS instead of OUTGOING_STATUS to match the documentation
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r167792 | mmichelson | 2009-01-08 13:48:42 -0600 (Thu, 08 Jan 2009) | 15 lines
Add the average talk time for a queue
This patch adds the functionality to app_queue of calculating
the average amount of time that channels are bridged for a
queue. The algorithm used to calculate the average is the same
exponential average currently used to calculate the average holdtime.
See the CHANGES file to see the methods you may use to view this
information.
(closes issue #13960)
Reported by: coolmig
Patches:
app_queue.c.diff.trunk-r158840 uploaded by coolmig (license 621)
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r167888 | mmichelson | 2009-01-08 16:34:52 -0600 (Thu, 08 Jan 2009) | 4 lines
Revert chan_sip changes which were accidentally committed
in revision 167792
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r167700 | kpfleming | 2009-01-08 10:43:26 -0600 (Thu, 08 Jan 2009) | 12 lines
Merged revisions 167620 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r167620 | kpfleming | 2009-01-07 17:32:21 -0600 (Wed, 07 Jan 2009) | 5 lines
When a SIP request or response arrives for a dialog with an associated Asterisk channel, and the lock on that channel cannot be obtained because it is held by another thread, instead of dropping the request/response, queue it for later processing when the channel lock becomes available.
http://reviewboard.digium.com/r/123/
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r167477 | lmadsen | 2009-01-07 13:18:45 -0500 (Wed, 07 Jan 2009) | 8 lines
Update queues.conf.sample documentation.
Update the queues.conf.sample documentation to mention that you need to preload chan_local.so as well if you plan on using Local channels for queue members, and you're preloading pbx_config.so.
(closes issue #14179)
Reported by: CrashHD
Tested by: CrashHD
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r167180 | mmichelson | 2009-01-05 10:59:36 -0600 (Mon, 05 Jan 2009) | 49 lines
Merged revisions 167179 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r167179 | mmichelson | 2009-01-05 10:51:59 -0600 (Mon, 05 Jan 2009) | 41 lines
A couple of changes to T.38 SDP attribute handling
There are some boolean attributes for T.38 such
as T38FaxFillBitRemoval, T38FaxTranscodingMMR, and
T38FaxTranscodingJBIG. By simply being present, we
should treat these as a "true" value. The current
code, however, was requiring a 1 or 0 as the value
of the attribute in order to parse it. This is due
to the fact that there are some T.38 endpoints and
gateways that also transmit this information
incorrectly. This patch follows the "be liberal in
what you accept and strict in what you send"
philosophy by accepting both the correctly- and
incorrectly-formatted attributes, but only sending
information as it is supposed to be sent.
It was also discovered that a particular type of
T.38 gateway sends some non-standard T.38 SDP
attributes. Instead of using T38FaxMaxDatagram
and T38MaxBitRate, it used T38MaxDatagram and
T38FaxMaxRate respectively. We now will properly
accept these attributes as well.
Note that there are a lot of patches cited in
the below commit message template. This is
because the person who submitted these patches is
an awesome person and wrote 1.4, 1.6.0, and 1.6.1
variants.
(closes issue #13976)
Reported by: linulin
Patches:
chan_sip.c.1.4-update1.diff uploaded by arcivanov (license 648)
chan_sip.c.1.6.0-update1.diff uploaded by arcivanov (license 648)
chan_sip.c.1.6.1-update1.diff uploaded by arcivanov (license 648)
chan_sip.c.1.4-relaxedT38_update1.diff uploaded by arcivanov (license 648)
chan_sip.c.1.6.0-relaxedT38_update1.diff uploaded by arcivanov (license 648)
chan_sip.c.1.6.1-relaxedT38_update1.diff uploaded by arcivanov (license 648)
Tested by: arcivanov
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r167125 | jpeeler | 2009-01-03 14:29:54 -0600 (Sat, 03 Jan 2009) | 3 lines
When parsing environment variable ASTERISK_PROMPT, make sure to proceed to the next character when a non format specifier is used (no %). Otherwise, the while loop looking for the null byte will never exit.
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r167021 | mmichelson | 2008-12-31 15:52:02 -0600 (Wed, 31 Dec 2008) | 4 lines
Change some incorrect syntax for pri set debug and correct
an off-by-one error in ss7 set debug command
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r166861 | mmichelson | 2008-12-29 12:04:52 -0600 (Mon, 29 Dec 2008) | 14 lines
Update app_queue to deal with the removal of AST_PBX_KEEPALIVE
When placing a call to a queue which ran a gosub on the member's
channel, Asterisk would crash every time, stemming from the fact
that the member's channel was being hung up unexpectedly when the
Gosub completed. The necessary change was pretty much copied and
pasted from app_dial's similar changes made last week.
I also took the opportunity to change a LOG_DEBUG message in
app_dial to use ast_debug. I am guessing this was due to a direct
merge from 1.4 that was not corrected to use trunk's preferred
syntax.
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r166823 | eliel | 2008-12-28 10:36:25 -0500 (Sun, 28 Dec 2008) | 3 lines
Fix a typo in the XML documentation of the AUDIOHOOK_INHERIT dialplan function.
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