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a peer reference is not leaked.
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and "tlsbindaddr".
Note: I don't think we can start properly without UDP port open, that needs to be tested.
- Removing "bindport" from configuration example, not needed to mention this any more
I suggest we deprecate "bindaddr" and "bindport" in trunk (for 1.6.1)
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- Adding IP address for TCP and/or TLS too if auto-domain is enabled and
binding to a different IP address
- Fixing documentation in sip.conf.sample
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- Set global_outboundproxy.force at reload.
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- Doxygen changes
- Replacing a doxygen description that was copied from another function
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come at a later stage,
as indicated by previous commit message
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problem with this
code, but not the correct problem. Thanks, anyway! ;-)
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sometime in the
future, but also want to avoid compiler warnings now.
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...trying to get my head around the thoughts behind the TCP/TLS stuff
and figure out what needs to be done to make it useful...
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TLS needs to have one configuration per configured domain at some point.
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Actually, kill the in-memory structure for type=user and start using the sip_peer
structure for every object. Have only one in-memory list and use them different
ways depending on type=user, type=peer and type=friend - like always.
The idea with this first patch is that configurations should work as before.
Some additional features for realtime peers. By adding a type= field, you
can now have multiple functions.
Let's test this for a while. Won't be integrated in 1.6.0, only in trunk,
for testing.
There's propably more to clean up and simplify here. Help is welcome
and encouraged!
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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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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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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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