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r234055 | russell | 2009-12-09 17:35:24 -0600 (Wed, 09 Dec 2009) | 2 lines
Move an entry from CHANGES to UPGRADE.txt.
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r234053 | russell | 2009-12-09 17:30:48 -0600 (Wed, 09 Dec 2009) | 2 lines
Move an entry from CHANGES that should be in UPGRADE.txt.
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r232657 | dvossel | 2009-12-02 17:27:45 -0600 (Wed, 02 Dec 2009) | 6 lines
update CHANGES and UPGRADE.txt for early media behavior change between 1.6.1 and 1.6.2
(closes issue #16212)
Reported by: miki
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r222110 | kpfleming | 2009-10-05 14:45:00 -0500 (Mon, 05 Oct 2009) | 25 lines
Allow non-compliant T.38 endpoints to be supportable via configuration option.
Many T.38 endpoints incorrectly send the maximum IFP frame size they can accept
as the T38FaxMaxDatagram value in their SDP, when in fact this value is
supposed to be the maximum UDPTL payload size (datagram size) they can accept.
If the value they supply is small enough (a commonly supplied value is '72'),
T.38 UDPTL transmissions will likely fail completely because the UDPTL packets
will not have enough room for a primary IFP frame and the redundancy used for
error correction. If this occurs, the Asterisk UDPTL stack will emit log messages
warning that data loss may occur, and that the value may need to be overridden.
This patch extends the 't38pt_udptl' configuration option in sip.conf to allow
the administrator to override the value supplied by the remote endpoint and
supply a value that allows T.38 FAX transmissions to be successful with that
endpoint. In addition, in any SIP call where the override takes effect, a debug
message will be printed to that effect. This patch also removes the
T38FaxMaxDatagram configuration option from udptl.conf.sample, since it has not
actually had any effect for a number of releases.
In addition, this patch cleans up the T.38 documentation in sip.conf.sample
(which incorrectly documented that T.38 support was passthrough only).
(issue #15586)
Reported by: globalnetinc
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r221592 | kpfleming | 2009-10-01 11:16:09 -0500 (Thu, 01 Oct 2009) | 12 lines
Remove ability to control T.38 FAX error correction from udptl.conf.
chan_sip has had the ability to control T.38 FAX error correction mode on a per-peer
(or global) basis for a couple of releases now, which is where it should have been
all along. This patch removes the ability to configure it in udptl.conf, but issues
a warning if the user tries to do, telling them to look at sip.conf.sample for how
to configure it now. For any SIP peers that are T.38 enabled in sip.conf, there is
already a default for FEC error correction even if the user does not specify any mode,
so this change will not turn off error correction by default, it will have the same
default value that has been in the udptl.conf sample file.
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r218799 | russell | 2009-09-16 08:34:41 -0500 (Wed, 16 Sep 2009) | 16 lines
Merged revisions 218798 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r218798 | russell | 2009-09-16 08:33:43 -0500 (Wed, 16 Sep 2009) | 9 lines
Remove the IAXy firmware from Asterisk.
The firmware can now be found on downloads.digium.com, where the rest of our
binary downloads live. This was the last part of our Asterisk tarballs that
was considered non-free by Debian. :-)
(closes issue #15838)
Reported by: paravoid
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r216092 | russell | 2009-09-03 14:38:35 -0500 (Thu, 03 Sep 2009) | 16 lines
Merged revisions 216085 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r216085 | russell | 2009-09-03 14:36:46 -0500 (Thu, 03 Sep 2009) | 9 lines
Merged revisions 216080 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r216080 | russell | 2009-09-03 14:35:23 -0500 (Thu, 03 Sep 2009) | 2 lines
Add a note about IAX2 to UPGRADE.txt.
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r210190 | kpfleming | 2009-08-03 15:48:48 -0500 (Mon, 03 Aug 2009) | 11 lines
Rename 'canreinvite' option to 'directmedia', with backwards compatibility.
It is clear from multiple mailing list, forum, wiki and other sorts of posts
that users don't really understand the effects that the 'canreinvite' config
option actually has, and that in some cases they think that setting it to 'no'
will actually cause various other features (T.38, MOH, etc.) to not work properly,
when in fact this is not the case. This patch changes the proper name of the
option to what it should have been from the beginning ('directmedia'), but
preserves backwards compatibility for existing configurations.
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r208464 | kpfleming | 2009-07-23 16:57:24 -0500 (Thu, 23 Jul 2009) | 46 lines
Rework of T.38 negotiation and UDPTL API to address interoperability problems
Over the past couple of months, a number of issues with Asterisk
negotiating (and successfully completing) T.38 sessions with various
endpoints have been found. This patch attempts to address many of
them, primarily focused around ensuring that the endpoints'
MaxDatagram size is honored, and in addition by ensuring that T.38
session parameter negotiation is performed correctly according to the
ITU T.38 Recommendation.
The major changes here are:
1) T.38 applications in Asterisk (app_fax) only generate/receive IFP
packets, they do not ever work with UDPTL packets. As a result of
this, they cannot be allowed to generate packets that would overflow
the other endpoints' MaxDatagram size after the UDPTL stack adds any
error correction information. With this patch, the application is told
the maximum *IFP* size it can generate, based on a calculation using
the far end MaxDatagram size and the active error correction mode on
the T.38 session. The same is true for sending *our* MaxDatagram size
to the remote endpoint; it is computed from the value that the
application says it can accept (for a single IFP packet) combined with
the active error correction mode.
2) All treatment of T.38 session parameters as 'capabilities' in
chan_sip has been removed; these parameters are not at all like
audio/video stream capabilities. There are strict rules to follow for
computing an answer to a T.38 offer, and chan_sip now follows those
rules, using the desired parameters from the application (or channel)
that wants to accept the T.38 negotiation.
3) chan_sip now stores and forwards ast_control_t38_parameters
structures for tracking 'our' and 'their' T.38 session parameters;
this greatly simplifies negotiation, especially for pass-through
calls.
4) Since T.38 negotiation without specifying parameters or receiving
the final negotiated parameters is not very worthwhile, the
AST_CONTROL_T38 control frame has been removed. A note has been added
to UPGRADE.txt about this removal, since any out-of-tree applications
that use it will no longer function properly until they are upgraded
to use AST_CONTROL_T38_PARAMETERS.
Review: https://reviewboard.asterisk.org/r/310/
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https://origsvn.digium.com/svn/asterisk/trunk
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r207224 | tilghman | 2009-07-17 17:04:43 -0500 (Fri, 17 Jul 2009) | 2 lines
Document the "flag" field in the voicemessages table.
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r204563 | tilghman | 2009-06-30 15:41:04 -0500 (Tue, 30 Jun 2009) | 13 lines
Merged revisions 204556 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r204556 | tilghman | 2009-06-30 15:23:51 -0500 (Tue, 30 Jun 2009) | 6 lines
More incorrect language codes, plus ensuring that regionalizations use the specified language, and not English for grammar.
(closes issue #15022)
Reported by: greenfieldtech
Patches:
20090519__issue15022.diff.txt uploaded by tilghman (license 14)
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r204470 | tilghman | 2009-06-30 13:36:24 -0500 (Tue, 30 Jun 2009) | 18 lines
Recorded merge of revisions 204469 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r204469 | tilghman | 2009-06-30 13:23:35 -0500 (Tue, 30 Jun 2009) | 11 lines
"tw" is the language specification for Twi (from Ghana) not Taiwanese.
(closes issue #15346)
Reported by: volivier
Patches:
20090617__issue15346__1.4.diff.txt uploaded by tilghman (license 14)
20090617__issue15346__trunk.diff.txt uploaded by tilghman (license 14)
20090617__issue15346__1.6.0.diff.txt uploaded by tilghman (license 14)
20090617__issue15346__1.6.1.diff.txt uploaded by tilghman (license 14)
20090617__issue15346__1.6.2.diff.txt uploaded by tilghman (license 14)
Tested by: volivier
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This patch includes a number of changes to the indications API. The primary
motivation for this work was to improve stability. The object management
in this API was significantly flawed, and a number of trivial situations could
cause crashes.
The changes included are:
1) Remove the module res_indications. This included the critical functionality
that actually loaded the indications configuration. I have seen many people
have Asterisk problems because they accidentally did not have an
indications.conf present and loaded. Now, this code is in the core,
and Asterisk will fail to start without indications configuration.
There was one part of res_indications, the dialplan applications, which did
belong in a module, and have been moved to a new module, app_playtones.
2) Object management has been significantly changed. Tone zones are now
managed using astobj2, and it is no longer possible to crash Asterisk by
issuing a reload that destroys tone zones while they are in use.
3) The API documentation has been filled out.
4) The API has been updated to follow our naming conventions.
5) Various bits of code throughout the tree have been updated to account
for the API update.
6) Configuration parsing has been mostly re-written.
7) "Code cleanup"
The code is from svn/asterisk/team/russell/indications/.
Review: http://reviewboard.digium.com/r/149/
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because SPRINTF() use non-literal format strings (which cannot be checked), move it into its own module so the rest of func_strings can benefit from format string checking
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indicating the callerid number just like the rest of asterisk.
(closes issue #13883)
Reported by: davidw
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previous branches
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ast_agi_unregister_multiple() and ast_agi_fdprintf() API calls to be consistent with API guidelines
also, move UPGRADE.txt to UPGRADE-1.6.txt and make the new UPGRADE.txt contain information about upgrading between Asterisk 1.6 releases
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hashing used by app_queue.c to be case-insensitive.
This is accomplished by adding a new case-insensitive
hashing function.
This was necessary to prevent bad refcount errors
(and potential crashes) which would occur due to the
fact that queues were initially read from the config
file in a case-sensitive manner. Then, when a user
issued a CLI command or manager action, we allowed
for case-insensitive input and used that input to
directly try to find the queue in the hash table. The result
was either that we could not find a queue that was input or
worse, we would end up hashing to a completely bogus value
based on the input.
This commit resolves the problem presented in
issue #13703. However, that issue was reported against
1.6.0. Since this fix introduces a behavior change, I am
electing to not place this same fix in to the 1.6.0 or 1.6.1
branches, and instead will opt for a change which does not
change behavior.
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devices and lines.
(closes issue #13412)
Reported by: wedhorn
Patches:
config-restruct-v4.diff uploaded by wedhorn (license 30)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@150426 f38db490-d61c-443f-a65b-d21fe96a405b
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background, by using the startup flag '-B'.
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r137530 | kpfleming | 2008-08-13 15:46:19 -0500 (Wed, 13 Aug 2008) | 1 line
add document describing what users will need to be aware of when upgrading to this version and using DAHDI
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column is actually there. If it's not, we still log everything else as
before.
(closes issue #13281)
Reported by: falves11
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(closes issue #13134)
Reported by: eliel
Patches:
app_image.c.patch uploaded by eliel (license 64)
UPGRADE.patch uploaded by eliel (license 64)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@134088 f38db490-d61c-443f-a65b-d21fe96a405b
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deprecation, of course)
(closes issue #13016)
Reported by: caio1982
Patches:
dialplan_globals6.diff uploaded by caio1982 (license 22)
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implementation from using the volatile libtds, to using the db-lib front end.
The unintended side effect of this is that we support (at least) versions 0.62
through 0.82 of the FreeTDS distribution without any #ifdef ugliness.
(closes issue #12844)
Reported by: jcollie
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function and begin the transition from SIPCHANINFO() to just using CHANNEL().
(closes issue #12856)
Reported by: mostyn
Patches:
iax_and_sip_channel_info.patch uploaded by mostyn (license 398)
(with some additional cleanup by me)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122802 f38db490-d61c-443f-a65b-d21fe96a405b
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(Closes issue #12831)
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(Inspired by a post on the -users list)
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argument instead
of the string "Urgent"
(closes issue #12660)
Reported by: jaroth
Patches:
externnotify.patch uploaded by jaroth (license 50)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@116592 f38db490-d61c-443f-a65b-d21fe96a405b
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marked "urgent" are considered to be higher priority than other messages
and so they will be played before any other messages in a user's mailbox.
There are two ways to leave an urgent message.
1. send the 'U' option to VoiceMail().
2. Set review=yes in voicemail.conf. This will give instructions for
a caller to mark a message as urgent after the message has been recorded.
I have tested that this works correctly with file and ODBC storage, and James
Rothenberger (who wrote initial support for this feature) has tested its use
with IMAP storage.
(closes issue #11817)
Reported by: jaroth
Based on branch http://svn.digium.com/svn/asterisk/team/jrothenberger/asterisk-urgent
Tested by: putnopvut, jaroth
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for the sake of backwards compatibility, since it is a non-trivial task to convert
existing large dialplans that depend on Macro() to use GoSub(), instead.
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which says the person's name is handled inside app_voicemail now.
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r111126 | kpfleming | 2008-03-26 14:51:24 -0500 (Wed, 26 Mar 2008) | 10 lines
Merged revisions 111125 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r111125 | kpfleming | 2008-03-26 14:49:30 -0500 (Wed, 26 Mar 2008) | 2 lines
update UPGRADE notes to document usage of the script
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r110962 | kpfleming | 2008-03-26 12:43:02 -0500 (Wed, 26 Mar 2008) | 2 lines
add note that the user will need to enable codec_ilbc to get it to build
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r110880 | kpfleming | 2008-03-26 09:42:35 -0700 (Wed, 26 Mar 2008) | 10 lines
Merged revisions 110869 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r110869 | kpfleming | 2008-03-26 08:53:46 -0700 (Wed, 26 Mar 2008) | 2 lines
due to licensing restrictions, we cannot distribute the source code for iLBC encoding and decoding... so remove it, and add instructions on how the user can obtain it themselves
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(closes issue #11236)
Reported by: philipps
Patches:
20080218__bug11236.diff.txt uploaded by Corydon76 (license 14)
Tested by: philipps
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@106072 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #12060)
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privilege to call out to a subshell.
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application. This will either be set to NOCHANNEL if the given channel was not found or SUCCESS if it worked.
(closes issue #11553)
Reported by: johan
Patches:
UPGRADE.txt.channelredirect.patch uploaded by johan (license 334)
CHANGES.channelredirect.patch uploaded by johan (license 334)
app_channelredirect-20080219.patch uploaded by johan (license 334)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103819 f38db490-d61c-443f-a65b-d21fe96a405b
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greater than two minutes). Remove
the saying of less-than for holdtime announcements since it can lead to awkward holdtime announcements. Using
'1' as a queue-round-seconds value is no longer valid.
(closes issue #9736)
Reported by: caio1982
Patches:
queue_announce5.diff uploaded by caio1982 (license 22)
Tested by: caio1982, putnopvut
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103687 f38db490-d61c-443f-a65b-d21fe96a405b
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2. Add a duration parameter to MusicOnHold
(closes issue #11904)
Reported by: dimas
Patches:
v2-moh.patch uploaded by dimas (license 88)
Tested by: dimas
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103658 f38db490-d61c-443f-a65b-d21fe96a405b
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This way, if people need to go back and review what was deprecated in previous
major releases, it is readily available to them. Thanks for the suggestion!
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103044 f38db490-d61c-443f-a65b-d21fe96a405b
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