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(if we have RTCP reports
and not use the p2p rtp bridge). I could not find a way to detect us using the p2p bridge, which
would be nice.
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file...
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application in the dialplan. Also minor documentation update explaining how to setup multiple signalling links within a linkset
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the IANA list
- add a few new registered and well-known extension names
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r127663 | murf | 2008-07-02 18:16:25 -0600 (Wed, 02 Jul 2008) | 30 lines
The CDRfix4/5/6 omnibus cdr fixes.
(closes issue #10927)
Reported by: murf
Tested by: murf, deeperror
(closes issue #12907)
Reported by: falves11
Tested by: murf, falves11
(closes issue #11849)
Reported by: greyvoip
As to 11849, I think these changes fix the core problems
brought up in that bug, but perhaps not the more global
problems created by the limitations of CDR's themselves
not being oriented around transfers.
Reopen if necc, but bug reports are not the best
medium for enhancement discussions. We need to start
a second-generation CDR standardization effort to cover
transfers.
(closes issue #11093)
Reported by: rossbeer
Tested by: greyvoip, murf
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May fix issue 12919.
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should not have session-timer
require headers, like MESSAGE and REFER. So in the future, only add them on requests and responses
that are related to INVITEs and re-INVITEs.
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It's mixing peers and users in a strange way and should really not be a CLI command,
since it's not meant for human output. It should be done with an app connecting to
manager.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r127560 | mmichelson | 2008-07-02 15:47:38 -0500 (Wed, 02 Jul 2008) | 3 lines
Fix thread-safety of some of the pbx_builtin_getvar_helper calls
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acceptdtmf and enddtmf. These allow for the DTMF pressed to be configurable
instead of being hardcoded to '#' and '*'.
(AST-86)
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We can still send video, just don't try to use what is not available.
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specific configuration in oss.conf
(reported by Klaus Darillion on the -video mailing list).
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recommended in RFC 3261, instead of being hardcoded to 32 seconds. This is
important for LANs, as it allows autocongestion to occur much more quickly, if
desired by the local PBX administrator. It also corrects a bug: if the T1
timer was increased beyond 500ms, then timer B would have been set at a much
lower value than recommended.
(closes issue #12544)
Reported by: kactus
Patches:
20080616__bug12544.diff.txt uploaded by Corydon76 (license 14)
Tested by: kactus
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without it being defined by each sip user.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r127133 | tilghman | 2008-07-01 15:25:37 -0500 (Tue, 01 Jul 2008) | 2 lines
Disable the old, slow search for matching callno in chan_iax2 (but allow it to be reenabled for debugging)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r127068 | tilghman | 2008-07-01 13:52:53 -0500 (Tue, 01 Jul 2008) | 8 lines
Change around how we schedule pings and lagrqs, and fix a reason why the
jobs were not getting properly cancelled.
(closes issue #12903)
Reported by: stevedavies
Patches:
20080620__bug12903__2.diff.txt uploaded by Corydon76 (license 14)
Tested by: stevedavies
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r126999 | tilghman | 2008-07-01 11:50:46 -0500 (Tue, 01 Jul 2008) | 2 lines
Suppress annoying warning by finding the remaining cases where the callno is not in the hash.
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r126902 | oej | 2008-07-01 16:59:31 +0200 (Tis, 01 Jul 2008) | 7 lines
Use domain part of SIP uri in register= configuration as fromdomain.
Reported by: one47
Patches:
sip-reg-fromdom2.dpatch uploaded by one47 (license 23)
(closes issue #12474)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r126899 | oej | 2008-07-01 16:27:33 +0200 (Tis, 01 Jul 2008) | 8 lines
Handle escaped URI's in call pickups. Patch by oej and IgorG.
Reported by: IgorG
Patches:
bug12299-11062-v2.patch uploaded by IgorG (license 20)
Tested by: IgorG, oej
(closes issue #12299)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r126789 | oej | 2008-07-01 13:51:38 +0200 (Tis, 01 Jul 2008) | 6 lines
Report 200 OK to all in-dialog OPTIONs requests (to confirm that the dialog
exist). Don't bother checking the request URI.
(closes issue #11264)
Reported by: ibc
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r126735 | oej | 2008-07-01 09:49:15 +0200 (Tis, 01 Jul 2008) | 7 lines
Fix bad XML for hold notification.
Reported by: gowen72
Patches:
hold.patch uploaded by gowen72 (license 432)
(closes issue #12942)
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fix a bug which caused a crash when a videodevice was
specified after startgui=1 in the config file. This also
involves a slightly different method to determine if the
gui is active or not.
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Merged revisions 126516 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r126516 | oej | 2008-06-30 14:50:55 +0200 (MÃ¥n, 30 Jun 2008) | 10 lines
Send all responses to an INVITE reliably, so that we retransmit if we don't get an ACK and
also fail if we don't get the very same precious ACK. Based on patch by tsearle, with
my own additions.
(closes issue #12951)
Reported by: tsearle
Patches:
busy_retransmit.patch uploaded by tsearle (license 373)
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giving you support for up to 9 video sources (e.g. webcams,
or X11 grabbers, etc.) active at once, displaying thumbnails for
each of them in the main GUI window, and with the ability to switch
between them on the fly during a conversation.
The code also implements a 'Picture in Picture' feature,
allowing you to select any source as primary or secondary,
and move the PiP window by just dragging it with the mouse.
The window looks like this:
________________________________________________________________
| ______ ______ ______ ______ ______ ______ ______ |
| | tn.1 | | tn.2 | | tn.3 | | tn.4 | | tn.5 | | tn.6 | | tn.7 | |
| |______| |______| |______| |______| |______| |______| |______| |
| ______ ______ ______ ______ ______ ______ ______ |
| |______| |______| |______| |______| |______| |______| |______| |
| _________________ __________________ _________________ |
| | | | | | | |
| | | | | | | |
| | | | | | | |
| | remote video | | | | local video | |
| | | | | | ______ | |
| | | | keypad | | | PIP || |
| | | | | | |______|| |
| |_________________| | | |_________________| |
| | | |
| | | |
| |__________________| |
|________________________________________________________________|
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some whitespace fixes
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the video source switching support
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you have multiple active instances of this grabber;
require v4l device names to start with '/dev/' - prevents some useless
attempt to open a file as a device.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r126056 | tilghman | 2008-06-27 17:01:09 -0500 (Fri, 27 Jun 2008) | 4 lines
When we get a 408 Timeout, don't stop trying to re-register.
(closes issue #12863)
Reported by: ricvil
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conversion.
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now take multiple arguments, the first one listed will be the one used
for new dialogs, and the rest listed will be acceptable ways for that peer to contact us. This fixes a minor bug where, because SIP TCP/UDP run on
the same port, could cause a TCP peer to be saved in the ast_db. There will also be warnings when a transport is changed for an unexpected reason.
(issue #12799)
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issue #12914)
Requested here: http://lists.digium.com/pipermail/asterisk-dev/2008-June/033658.html
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r125740 | tilghman | 2008-06-27 07:19:39 -0500 (Fri, 27 Jun 2008) | 7 lines
Add proper deadlock avoidance.
(closes issue #12914)
Reported by: ozan
Patches:
20080625__bug12914.diff.txt uploaded by Corydon76 (license 14)
Tested by: ozan
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This is now handled in skinny_register and load_config.
part two of chan_skinny cleanup
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This has never been used, and noone could give us info about what
it is used for.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r125384 | oej | 2008-06-26 18:32:08 +0200 (Tor, 26 Jun 2008) | 3 lines
Add support for peer realm based auth (a few missing lines, the rest is well documented but never worked)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r125327 | kpfleming | 2008-06-26 10:30:33 -0500 (Thu, 26 Jun 2008) | 5 lines
ensure that (whenever possible) if we generate a log message because an ioctl() call to DAHDI/Zaptel failed, that we include the reason it failed by including the stringified error number
(issue AST-80)
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