https://origsvn.digium.com/svn/asterisk/branches/1.4
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r130102 | tilghman | 2008-07-11 11:50:42 -0500 (Fri, 11 Jul 2008) | 9 lines
Pass the devicestate from an underlying channel up through the Agent channel.
This should make the Agent always report the correct device state, even when
the underlying channel is used for other purposes.
(closes issue #12773)
Reported by: davidw
Patches:
20080710__bug12773.diff.txt uploaded by Corydon76 (license 14)
Tested by: davidw
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r130042 | kpfleming | 2008-07-11 11:08:03 -0500 (Fri, 11 Jul 2008) | 5 lines
new installations should be using DAHDI instead of Zaptel, so the sample config file is now chan_dahdi.conf instead of zapata.conf
also, convert remaining references to zapata.conf in various places
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r130039 | kpfleming | 2008-07-11 10:41:56 -0500 (Fri, 11 Jul 2008) | 4 lines
add support for a configuration parameter for 'inband audio during RELEASE', which is currently mandatory in libpri-1.4.4 but will become configurable in libpri-1.4.5 later today
(related to issue #13042)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r129966 | kpfleming | 2008-07-11 09:03:52 -0500 (Fri, 11 Jul 2008) | 5 lines
fix a flaw found while experimenting with structure alignment and padding; low-fence checking would not work properly on 64-bit platforms, because the compiler was putting 4 bytes of padding between the fence field and the allocation memory block
added a very obvious runtime warning if this condition reoccurs, so the developer who broke it can be chastised into fixing it :-)
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r129967 | kpfleming | 2008-07-11 09:03:52 -0500 (Fri, 11 Jul 2008) | 5 lines
simplify calculation
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fn2 was used in three functions. In every case, it was initialized
in the function it was used in. This meant there was no need
to have it in a malloc'd structure just taking up space. Furthermore
two of the functions it was used in were completely unnecessary since
fn2 was set to exactly the same value as the vm_state's fn string.
fn2 was a char array sized at PATH_MAX. On my system, PATH_MAX is
4096. This equates to a 4K memory savings per vm_state allocated.
Since there is a vm_state malloc'd for every voicemail user on
the system, this could potentially add up nicely if there are lots
of users. In addition, a vm_state is allocated on the stack each
time a caller calls the VoiceMailMain application, meaning that
there is a significant stack savings with this patch too.
Of course, a single vm_state struct still takes up approximately
20K on my system (when using IMAP storage. Without IMAP storage,
there would be about another 300 bytes fewer usage), even with
this removal. Further optimizations are probably possible,
but most likely not as easy as this one.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129734 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r129436 | mmichelson | 2008-07-09 14:32:20 -0500 (Wed, 09 Jul 2008) | 13 lines
Fix a problem where inbound rfc2833 audio would be sent to the
core instead of being P2P bridged. When the core regenerated
the rfc2833 packet for the outbound leg, the SSRC would be different
than the RTP audio on the call leg causing DTMF detection issues on
the far end.
(closes issue #12955)
Reported by: tonyredstone
Patches:
dynamic_rtp.patch uploaded by tsearle (license 373)
Tested by: tonyredstone
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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
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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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
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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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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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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128524 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Tilghman Lesher
Kevin P. Fleming
Russell Bryant
Sean Bright
Terry Wilson
Mark Michelson
Brett Bryant
Michiel van Baak
Matthew Fredrickson
Olle Johansson