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r129734 | mmichelson | 2008-07-10 15:33:13 -0500 (Thu, 10 Jul 2008) | 23 lines
Removed the fn2 field from the vm_state structure.
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.
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r129684 | bbryant | 2008-07-10 14:13:12 -0500 (Thu, 10 Jul 2008) | 8 lines
Fixes a bug where the interface for a queue member gets reloaded as the state_interface, if a state_interface was set, on reload because the
state_interface isn't stored in the ast_db.
(closes issue #13043)
Reported by: jvandal
Patches:
app_queue.patch uploaded by jvandal (license 413)
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r129437 | mmichelson | 2008-07-09 14:40:30 -0500 (Wed, 09 Jul 2008) | 21 lines
Merged revisions 129436 via svnmerge from
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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r129159 | mmichelson | 2008-07-08 16:10:33 -0500 (Tue, 08 Jul 2008) | 21 lines
Blocked revisions 129158 via svnmerge
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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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r129156 | bbryant | 2008-07-08 16:00:01 -0500 (Tue, 08 Jul 2008) | 6 lines
Fix a bug in SRV lookups where dnsmgr would discard everything but the
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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r129006 | russell | 2008-07-08 09:17:37 -0500 (Tue, 08 Jul 2008) | 9 lines
Update app_fax for better compatibility with spandsp 0.0.5. Add a call to
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)
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r128951 | oej | 2008-07-08 12:02:12 +0200 (Tis, 08 Jul 2008) | 19 lines
Merged revisions 128950 via svnmerge from
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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Blocked revisions 128197 via svnmerge
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r128197 | oej | 2008-07-05 21:27:42 +0200 (Lör, 05 Jul 2008) | 5 lines
Add new SIP cli command "sip show channelstats" that displays some QoS data (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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r128731 | mmichelson | 2008-07-07 15:28:33 -0500 (Mon, 07 Jul 2008) | 7 lines
If imapfolder=foo were set in voicemail.conf, then when calling VoiceMailMain,
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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r128640 | mmichelson | 2008-07-07 12:09:11 -0500 (Mon, 07 Jul 2008) | 18 lines
Merged revisions 128639 via svnmerge from
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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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r128564 | russell | 2008-07-07 06:53:52 -0500 (Mon, 07 Jul 2008) | 3 lines
As pointed out on the -dev list, actually use the result of find_peer() so that
a peer reference is not leaked.
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r128525 | oej | 2008-07-06 22:19:04 +0200 (Sön, 06 Jul 2008) | 7 lines
- Adding alias "udpbindaddr" for the UDP port to comply with "tcpbindaddr" 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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r128524 | oej | 2008-07-06 22:11:37 +0200 (Sön, 06 Jul 2008) | 5 lines
- Fixing issues with "sip show settings"
- 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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in trunk.
Merged revisions 128417 via svnmerge from
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r128417 | oej | 2008-07-06 12:13:45 +0200 (Sön, 06 Jul 2008) | 3 lines
Adding documentation on the T.140 support in Asterisk. This is a function that we're
the reference implementation on now. :-)
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r128381 | oej | 2008-07-06 10:43:08 +0200 (Sön, 06 Jul 2008) | 3 lines
Remove comments that doesn't make sense. The deprecation of type=user will come at a later stage,
as indicated by previous commit message
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r128379 | oej | 2008-07-06 10:32:11 +0200 (Sön, 06 Jul 2008) | 4 lines
Fix severe problem with my previous commit of "kill-the-user". Russell saw a problem with this
code, but not the correct problem. Thanks, anyway! ;-)
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r128378 | oej | 2008-07-06 10:28:58 +0200 (Sön, 06 Jul 2008) | 3 lines
Changing name of global api call to ast_*
My mistake, pointed out by Russell.
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r128290 | oej | 2008-07-05 23:55:57 +0200 (Lör, 05 Jul 2008) | 5 lines
Adding doxygen comments to missing parts, moving some #define
...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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