Move phoneprov config files to be installed with 'make samples' so changes aren't inadvertently lost on a 'make install'


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129738 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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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(closes issue #13038)
Reported by: eliel
Patches:
      finish.deprecate.patch uploaded by eliel (license 64)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129581 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

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r129567 | russell | 2008-07-10 11:03:59 -0500 (Thu, 10 Jul 2008) | 3 lines

Note that pbx_spool.so is the module used for call files
(inspired by a question in #asterisk)

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r129505 | seanbright | 2008-07-10 09:57:19 -0400 (Thu, 10 Jul 2008) | 2 lines

Update svn:ignore

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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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https://origsvn.digium.com/svn/asterisk/branches/1.4

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r129343 | seanbright | 2008-07-09 09:41:21 -0400 (Wed, 09 Jul 2008) | 4 lines

Look for the system installed awk instead of assuming it's at /usr/bin/awk.

Pointed out by jmls via #asterisk-dev.

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129159 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129152 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129048 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(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

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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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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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128733 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128564 65c4cc65-6c06-0410-ace0-fbb531ad65f3

  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

- 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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