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first SRV result from DNS before processing weights and priorities and
dns_parse_answer wouldn't report that there were no records in DNS
unless a failure occured. Also fixed a bug where dnsmgr_refresh would
report that a entry was being changed when ast_gethostbyname had failed.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r129149 | tilghman | 2008-07-08 15:27:47 -0500 (Tue, 08 Jul 2008) | 8 lines
Cause SIP to return a 480 instead of a 404 when a sip peer exists, but is not
registered.
(closes issue #12885)
Reported by: ibc
Patches:
20080701__bug12885__2.diff.txt uploaded by Corydon76 (license 14)
Tested by: ibc
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(closes issue #13001)
Reported by: eliel
Patches:
20080708__bug13001.diff.txt uploaded by Corydon76 (license 14)
20080708__bug13001.diff.txt.fixed uploaded by eliel (license 64)
Tested by: Corydon76, eliel
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r129047 | tilghman | 2008-07-08 11:45:23 -0500 (Tue, 08 Jul 2008) | 7 lines
Timestamp decoding for video mini-frames is bogus, because the timestamp only
includes 15 bits, unlike voice frames, which contain a 16-bit timestamp.
(closes issue #13013)
Reported by: jpgrayson
Patches:
chan_iax2_unwrap_ts.patch uploaded by jpgrayson (license 492)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129048 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #13002)
Reported by: caio1982
Patches:
janitor_arraylen5.diff uploaded by caio1982 (license 22)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129045 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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t38_terminal_release, and make sure that the phase E handler gets called
with proper status.
(closes issue #13020)
Reported by: dimas
Patches:
v1-appfax.patch uploaded by dimas (license 88)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129006 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r128950 | oej | 2008-07-08 11:52:21 +0200 (Tis, 08 Jul 2008) | 11 lines
Don't hangup the call if we can't resolve the Contact if there's a proxy
route set for the call.
----
This comment was added a while ago and today it hit me badly.
/* OEJ: Possible issue that may need a check:
If we have a proxy route between us and the device,
should we care about resolving the contact
or should we just send it?
*/
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r128912 | oej | 2008-07-08 11:06:08 +0200 (Tis, 08 Jul 2008) | 7 lines
Fix issues where repeated messages where ignored, but retransmitted reliably instead of unreliably.
Reported by: johan
Patches:
12746.txt uploaded by oej (license 306)
Tested by: johan
(issue #12746)
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r128856 | tilghman | 2008-07-07 19:01:30 -0500 (Mon, 07 Jul 2008) | 7 lines
Check for non-NULL before stripping characters.
(closes issue #12954)
Reported by: bfsworks
Patches:
20080701__bug12954.diff.txt uploaded by Corydon76 (license 14)
Tested by: deti
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r128812 | tilghman | 2008-07-07 18:21:52 -0500 (Mon, 07 Jul 2008) | 2 lines
Stop using deprecated method, as requested by Kevin.
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r128795 | russell | 2008-07-07 17:41:48 -0500 (Mon, 07 Jul 2008) | 8 lines
Fix handling of when a pvt disappears. Properly return the pvt locked
and don't hold the pvt lock while destroying the ast_channel.
(closes issue #13014)
Reported by: jpgrayson
Patches:
chan_iax2_ast_iax2_new2.patch uploaded by jpgrayson (license 492)
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r128737 | seanbright | 2008-07-07 16:47:56 -0400 (Mon, 07 Jul 2008) | 9 lines
Remove spurious trailing whitespace from log messages and fix a spelling error
in a log message.
(closes issue #13017)
Reported by: jpgrayson
Patches:
chan_iax2_space_after_newline.patch uploaded by jpgrayson (license 492)
chan_iax2_spelling.patch uploaded by jpgrayson (license 492)
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app_voicemail would attempt to play a file called vm-foo instead of playing
vm-INBOX to play the "new" sound file. This commit fixes that issue.
This may fix one of the problems reported in issue #12987
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Brought up by reporter on issue #12987
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r128639 | mmichelson | 2008-07-07 12:02:28 -0500 (Mon, 07 Jul 2008) | 10 lines
By using the iaxdynamicthreadcount to identify a thread, it was possible
for thread identifiers to be duplicated. By using a globally-unique monotonically-
increasing integer, this is now avoided.
(closes issue #13009)
Reported by: jpgrayson
Patches:
chan_iax2_dyn_threadnum.patch uploaded by jpgrayson (license 492)
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r128637 | kpfleming | 2008-07-07 11:51:50 -0500 (Mon, 07 Jul 2008) | 9 lines
use tzafrir's patch to fix this problem properly... i made the previous set of changes without thoroughly testing them, doh!
(closes issue #12911)
Reported by: tzafrir
Patches:
custum_dahdi_configure_2.diff uploaded by tzafrir (license 46)
Tested by: tzafrir
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in the documentation.
Patch provided by caio1982 (license 22)
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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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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that we're
the reference implementation on now. :-)
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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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My mistake, pointed out by Russell.
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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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AMI commands can display that a channel is under control of an AGI.
Work inspired by work at customer site, but paid for by Edvina AB
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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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(closes issue #12860)
Reported by: flyn
Patches:
asterisk.ldif uploaded by suretec (license 70)
asterisk.schema uploaded by suretec (license 70)
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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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r128029 | tilghman | 2008-07-04 11:11:03 -0500 (Fri, 04 Jul 2008) | 2 lines
Move the free down one
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