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user guide
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.1.6@216102 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@208503 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/trunk
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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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https://origsvn.digium.com/svn/asterisk/trunk
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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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https://origsvn.digium.com/svn/asterisk/trunk
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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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https://origsvn.digium.com/svn/asterisk/trunk
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r179154 | russell | 2009-02-27 15:23:12 -0600 (Fri, 27 Feb 2009) | 2 lines
Add a note about the ordering of entries in sip.conf in 1.6.1.
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https://origsvn.digium.com/svn/asterisk/trunk
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r158924 | mnicholson | 2008-11-24 18:05:41 -0600 (Mon, 24 Nov 2008) | 6 lines
Make the Join event from app_queue use CallerIDNum insead of CallerID for
indicating the callerid number just like the rest of asterisk.
(closes issue #13883)
Reported by: davidw
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https://origsvn.digium.com/svn/asterisk/trunk
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r158449 | kpfleming | 2008-11-21 14:42:37 -0600 (Fri, 21 Nov 2008) | 3 lines
as suggested by jtodd, document the purposes of the CHANGES and UPGRADE files
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https://origsvn.digium.com/svn/asterisk/trunk
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r157706 | kpfleming | 2008-11-19 06:42:19 -0600 (Wed, 19 Nov 2008) | 5 lines
make some corrections to the ast_agi_register_multiple(), 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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@137203 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135717 f38db490-d61c-443f-a65b-d21fe96a405b
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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)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@131606 f38db490-d61c-443f-a65b-d21fe96a405b
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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
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@126226 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@124835 f38db490-d61c-443f-a65b-d21fe96a405b
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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)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@121855 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@120567 f38db490-d61c-443f-a65b-d21fe96a405b
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(Inspired by a post on the -users list)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118223 f38db490-d61c-443f-a65b-d21fe96a405b
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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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114931 f38db490-d61c-443f-a65b-d21fe96a405b
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which says the person's name is handled inside app_voicemail now.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114841 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@104110 f38db490-d61c-443f-a65b-d21fe96a405b
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privilege to call out to a subshell.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@104039 f38db490-d61c-443f-a65b-d21fe96a405b
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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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now present that can not be used at the same time as chan_alsa or chan_oss.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96715 f38db490-d61c-443f-a65b-d21fe96a405b
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- Refer to the proper documentation
- Implement separate signalling/media QoS/CoS in many channels using RTP
- Improve warnings and verbose messages
- Deprecate some old settings
Minor modifications by me, a big effort from IgorG.
Thanks!
Reported by: IgorG
Patches:
qoscleanup-89394-4-trunk.patch uploaded by IgorG (license 20)
Tested by: IgorG
(closes issue #11145)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93163 f38db490-d61c-443f-a65b-d21fe96a405b
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Reported by: eliel
Patch by: eliel
(Closes issue #11344)
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version 1.1 - hopefully a more consistent manager interface.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91438 f38db490-d61c-443f-a65b-d21fe96a405b
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"call" level.
(Closes issue #11015)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91173 f38db490-d61c-443f-a65b-d21fe96a405b
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"Username" still works, but is deprecated.
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- Restructure other changes to UPGRADE.txt and CHANGES
We're still looking for scripts that replace
asterisk -rx "show shannels concise"
by using the manager interface, but still produces the same output.
Anyone?
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89606 f38db490-d61c-443f-a65b-d21fe96a405b
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Also, some cleanup to CHANGES.
Reported by: michael-fig
Patch by: michael-fig,tilghman
(Closes issue #11004)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89291 f38db490-d61c-443f-a65b-d21fe96a405b
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Closes issue #10614
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QUEUE_MEMBER_COUNT function.
A deprecation notice will be issued the first time QUEUE_MEMBER_COUNT is used.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@87395 f38db490-d61c-443f-a65b-d21fe96a405b
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Remove old unused defines for old style.
Closes issue 10860, patch by IgorG.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85764 f38db490-d61c-443f-a65b-d21fe96a405b
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CLI-style output (closes issue #8254)
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