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r129208 | mmichelson | 2008-07-08 16:31:51 -0500 (Tue, 08 Jul 2008) | 3 lines
Update documentation to have the correct option name
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r129158 | mmichelson | 2008-07-08 16:09:10 -0500 (Tue, 08 Jul 2008) | 14 lines
Backport TCP-related timeouts to IMAP voicemail in 1.4
since it should solve bugs people are experiencing. Specifically,
there are times where communication with the IMAP server causes
system calls to block forever. If this should happen when querying
the mailbox so that chan_sip's do_monitor thread can send MWI to
a phone, it means that SIP calls cannot be processed any more.
The timeout options are outlined in doc/imapstorage.txt. Defaults
for the timeouts are sixty seconds.
(closes issue #12987)
Reported by: mthomasslo
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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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129156 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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(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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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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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