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A simple patch to enable auto tab complete.
(closes issue #17152)
Reported by: pabelanger
Patches:
0017152.patch uploaded by pabelanger (license 224)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256821 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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In troff '-' is used for a hyphen. A minus is denoted by '\-' . This is
normally also used for a dash.
This patch converts all '-'-s that are minuses or dashes to '\-'.
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Call completion code tries to grab the call completion parameters
from the requesting channel during local_request. When originating
a call to a local channel, however, this channel is NULL. This
was causing an issue for me when trying to run a test script.
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From the original issue report opened by Nick Lewis:
Many sip headers in many sip methods contain the ABNF structure
name-andor-addr = name-addr / addr-spec
Examples include the to-header, from-header, contact-header, replyto-header
At the moment chan_sip.c makes various different attempts to parse this name-andor-addr structure for each header type and for each sip method with sometimes limited degrees of success.
I recommend that this name-andor-addr structure be parsed by a dedicated function and that it be used irrespective of the specific method or header that contains the name-andor-addr structure
Nick has also included unit tests for verifying these routines as well, so...heck yeah.
(closes issue #16708)
Reported by: Nick_Lewis
Patches:
reqresp_parser-nameandoraddr2.patch uploaded by Nick Lewis (license 657
Review: https://reviewboard.asterisk.org/r/549
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From Reviewboard:
CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date
overview of the architecture can be found in the file doc/CCSS_architecture.pdf
in the CCSS branch. Off the top of my head, the big differences between what is
implemented and what is in the document are as follows:
1. We did not end up modifying the Hangup application at all.
2. The document states that a single call completion monitor may be used across
multiple calls to the same device. This proved to not be such a good idea
when implementing protocol-specific monitors, and so we ended up using one
monitor per-device per-call.
3. There are some configuration options which were conceived after the document
was written. These are documented in the ccss.conf.sample that is on this
review request.
For some basic understanding of terminology used throughout this code, see the
ccss.tex document that is on this review.
This implements CCBS and CCNR in several flavors.
First up is a "generic" implementation, which can work over any channel technology
provided that the channel technology can accurately report device state. Call
completion is requested using the dialplan application CallCompletionRequest and can
be canceled using CallCompletionCancel. Device state subscriptions are used in order
to monitor the state of called parties.
Next, there is a SIP-specific implementation of call completion. This method uses the
methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion
using SIP signaling. There are a few things to note here:
* The agent/monitor terminology used throughout Asterisk sometimes is the reverse of
what is defined in the referenced draft.
* Implementation of the draft required support for SIP PUBLISH. I attempted to write
this in a generic-enough fashion such that if someone were to want to write PUBLISH
support for other event packages, such as dialog-state or presence, most of the effort
would be in writing callbacks specific to the event package.
* A subportion of supporting PUBLISH reception was that we had to implement a PIDF
parser. The PIDF support added is a bit minimal. I first wrote a validation
routine to ensure that the PIDF document is formatted properly. The rest of the
PIDF reading is done in-line in the call-completion-specific PUBLISH-handling
code. In other words, while there is PIDF support here, it is not in any state
where it could easily be applied to other event packages as is.
Finally, there are a variety of ISDN-related call completion protocols supported. These
were written by Richard Mudgett, and as such I can't really say much about their
implementation. There are notes in the CHANGES file that indicate the ISDN protocols
over which call completion is supported.
Review: https://reviewboard.asterisk.org/r/523
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From Review Board:
There are two interrelated changes here.
First, there is the introduction of func_srv. This adds two new read-only
dialplan functions, SRVQUERY and SRVRESULT. They work very similarly to the
ENUMQUERY and ENUMRESULT functions, except that this allows one to query SRV
records instead. In order to facilitate this work, I added a couple of new API
calls to srv.h. ast_srv_get_record_count tells the number of records returned
by an SRV lookup. This number is calculated at the time of the SRV lookup.
ast_srv_get_nth_record allows one to get a numbered SRV record.
Second, there is the modification to chan_sip that allows one to specify a
hostname or IP address (along with a port) to send an outgoing INVITE to when
dialing a SIP peer. This goes hand-in-hand with func_srv. You can query SRV
records and then use the host and port from the results to dial via a specific
host instead of what is configured in sip.conf.
Review: https://reviewboard.asterisk.org/r/608
SWP-1200
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exist.
Using wildcard matching in the Makefile is not adequate to determine whether
an export file should exist for a module or not, so instead we'll just
create one if the module needs one, or copy the default one if it does not.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256428 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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test for this.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256370 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #16767)
Reported by: lmsteffan
Patches:
deadlock_16767v3.diff uploaded by dvossel (license 671)
Review: https://reviewboard.asterisk.org/r/606/
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r256225 | rmudgett | 2010-04-05 19:10:16 -0500 (Mon, 05 Apr 2010) | 5 lines
DAHDI/PRI call to pri_channel_bridge() not protected by PRI lock.
SWP-1231
ABE-2163
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SWP-1229
ABE-2161
* Ensure chan_local.c:local_call() will not leak cid.cid_dnid when
copying.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256104 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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crash.
Fix app_dial.c:do_forward() OPT_FORCECLID setting cid.cid_num with a stack
allocated string instead of a heap allocated string.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256103 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #16866)
Reported by: DEA
Patches:
rt-meetme-options.txt uploaded by DEA (license 3)
Tested by: DEA
Review: https://reviewboard.asterisk.org/r/582/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r256014 | russell | 2010-04-02 18:45:56 -0500 (Fri, 02 Apr 2010) | 9 lines
Resolve a deadlock that occurs due to a pointless call to ast_bridged_channel()
(closes issue #16840)
Reported by: bzing2
Patches:
patch.txt uploaded by bzing2 (license 902)
issue_16840.rev1.diff uploaded by russell (license 2)
Tested by: bzing2, russell
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r256009 | russell | 2010-04-02 18:30:15 -0500 (Fri, 02 Apr 2010) | 2 lines
Remove extremely verbose debug message.
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(closes issue #17065)
Reported by: globalnetinc
Patches:
astcanary.patch uploaded by makoto (license 38)
Tested by: frawd, globalnetinc
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255952 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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prefixes.
Some platforms prefix externally-visible symbols in object files generated
from C sources (most commonly, '_' is the prefix). On these platforms,
the existing symbol export filtering process ends up suppressing all the symbols
that are supposed to be left visible. This patch allows the prefix string
to be supplied to the top-level Makefile in the LINKER_SYMBOL_PREFIX variable,
and then generates the linker scripts as required to include the prefix
supplied.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255906 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #17126)
Reported by: wedhorn
Patches:
skinny79xx_redial1.diff uploaded by wedhorn (license 30)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255851 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Bulk lot of generally trivial changes for cleaning up the transmit stuff. Line state request has been modified for line only responses.
(closes issue #16994)
Reported by: wedhorn
Patches:
skinny-clean07.diff uploaded by wedhorn (license 30)
Tested by: wedhorn
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255850 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #16828)
Reported by: oej
Patches:
20100331__issue16828.diff.txt uploaded by tilghman (license 14)
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sip.conf
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There was a bug where we split the URI on the @ sign and then attempted
to compare to "anonymous@anonymous.invalid" afterwards. This comparison
could never evaluate true. So now we keep a copy of the URI prior to the
split so that the comparison is valid.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r255591 | tilghman | 2010-03-31 14:09:46 -0500 (Wed, 31 Mar 2010) | 15 lines
Ensure line terminators in email are consistent.
Fixes an issue with certain Mail Transport Agents, where attachments are not
interpreted correctly.
(closes issue #16557)
Reported by: jcovert
Patches:
20100308__issue16557__1.4.diff.txt uploaded by tilghman (license 14)
20100308__issue16557__1.6.0.diff.txt uploaded by tilghman (license 14)
20100308__issue16557__trunk.diff.txt uploaded by tilghman (license 14)
Tested by: ebroad, zktech
Reviewboard: https://reviewboard.asterisk.org/r/544/
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(closes issue #17021)
Reported by: kovzol
Tested by: lmadsen, kovzol, davidw, ebroad
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r255409 | russell | 2010-03-30 15:56:00 -0500 (Tue, 30 Mar 2010) | 2 lines
Don't kill Asterisk if the H323 listener does not start.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r255322 | russell | 2010-03-30 11:06:06 -0500 (Tue, 30 Mar 2010) | 2 lines
Don't make Asterisk not start if pbx_dundi fails to initialize.
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matching the devstate handling of the MeetMe conferences.
Review: https://reviewboard.asterisk.org/r/572/
Closes issue #16972
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This should give a good speed boost, in that one particular thread isn't waking
up once a second to read directory contents.
Reviewboard: https://reviewboard.asterisk.org/r/137/
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This documentation associated wth tlsbindaddr is still useful so lets
synchronize it between trunk and 1.6.x branches.
(issue #17054)
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Update some confusing documentation for the tlsbindaddr
option in sip.conf.sample. Point at a link instead which
has better documentation.
(closes issue #17054)
Reported by: klaus3000
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before shift'ing.
Reported and tested by pabelanger.
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Now that these files are in the tree, they should prefer the tree's local
copy of all Asterisk headers over any that may be installed.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r254800 | qwell | 2010-03-25 15:41:15 -0500 (Thu, 25 Mar 2010) | 1 line
Don't remove local copies of utils in uninstall.
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r254714 | qwell | 2010-03-25 14:39:23 -0500 (Thu, 25 Mar 2010) | 4 lines
Fix DEBUG_THREADS issue with out-of-tree modules.
Take 2, without ABI breakage this time.
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The ast_channel structure had a field used for linking a channel into a
linked list, but now that ast_channel structures are ao2 objects, this is
no longer needed, and could be harmful as ao2 objects really shouldn't
ever be placed into linked lists (since those lists don't assist with
reference count management on the objects).
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