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

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r110962 | kpfleming | 2008-03-26 12:43:02 -0500 (Wed, 26 Mar 2008) | 2 lines

add note that the user will need to enable codec_ilbc to get it to build

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110963 65c4cc65-6c06-0410-ace0-fbb531ad65f3

revert something dumb, because i was running svn diff in a subfolder not the root of trunk, before doing my commit and did not see it

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110930 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #12067)
Reported by: juggie
Patches:
      12067_snmp_doc.patch uploaded by juggie (license 24)
Tested by: juggie


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110911 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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r110880 | kpfleming | 2008-03-26 09:42:35 -0700 (Wed, 26 Mar 2008) | 10 lines

Merged revisions 110869 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r110869 | kpfleming | 2008-03-26 08:53:46 -0700 (Wed, 26 Mar 2008) | 2 lines

due to licensing restrictions, we cannot distribute the source code for iLBC encoding and decoding... so remove it, and add instructions on how the user can obtain it themselves

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110881 65c4cc65-6c06-0410-ace0-fbb531ad65f3

change, it was possible to start Asterisk with the manager interface enabled, then
either comment out the enabled option or make manager.conf unopenable and the manager
interface would still be enabled.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110831 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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r110779 | qwell | 2008-03-25 17:51:17 -0500 (Tue, 25 Mar 2008) | 6 lines

Make file access in cdr_custom similar to cdr_csv.

Fixes issue #12268.

Patch borrowed from r82344

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110780 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110726 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #11969)
 Reported by: pprindeville
 Patches: 
       acme-virtualpbx.1.6.patch uploaded by pprindeville (license 347)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110691 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #12293)
 Reported by: pprindeville
 Patches: 
       bugid-0012293.1.6.patch uploaded by pprindeville (license 347)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110689 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110639 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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r110635 | mmichelson | 2008-03-25 10:40:33 -0500 (Tue, 25 Mar 2008) | 7 lines

When reverting a commit, I accidentally left in this bit which was an experiment
to see what would happen. It passed the compile test, and I didn't notice I had
left this change in too.

So this is a revert of a revert...sort of.


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110636 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add a special dialplan variable to chan_sip which will cause an audio file to be played upon completion of an attended transfer.
(closes issue #9239)
Reported by: sunder


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110631 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4 lines

Add an option (transmit_silence) which transmits silence during both Record() and DTMF generation. The reason this is an option is that in order to transmit silence we have to setup a translation path. This may not be needed/wanted in all cases.
(closes issue #10058)
Reported by: tracinet

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110629 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #12278)
Reported by: rjain
Patches: 
      chan_sip.c.diff uploaded by rjain (license 226)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110625 65c4cc65-6c06-0410-ace0-fbb531ad65f3

SIP REGISTER requests. rjain points out that RFC 3265 specifies
that the Event: header is not a valid header for REGISTER requests
and that the "registration" value is not defined at IANA.

(closes issue #12279)
Reported by: rjain
Patches:
      chan_sip.c.diff uploaded by rjain (license 226)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110621 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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r110618 | mmichelson | 2008-03-24 14:17:41 -0500 (Mon, 24 Mar 2008) | 15 lines

This is a revert for revision 108288. The reason is that that revision
was not for an actual bug fix per se, and so it really should not have been in 1.4 in
the first place. Plus, people who compile with DO_CRASH are more likely
to encounter a crash due to this change. While I think the usage of DO_CRASH
in ast_sched_del is a bit absurd, this sort of change is beyond the scope of 1.4
and should be done instead in a developer branch based on trunk 
so that all scheduler functions are fixed at once.

I also am reverting the change to trunk and 1.6 since they also suffer from
the DO_CRASH potential.

(closes issue #12272)
Reported by: qq12345


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110619 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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r110614 | russell | 2008-03-24 12:34:56 -0500 (Mon, 24 Mar 2008) | 2 lines

Turn a NOTICE into a DEBUG message.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110615 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Only print out the set_address_from_contact host verbose message if debugging is enabled on the dialog.
(closes issue #12280)
Reported by: rjain
Patches:
      chan_sip.c.diff uploaded by rjain (license 226)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110610 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110578 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merge over ast_audiohook_volume branch. This adds API calls for use by developers to adjust the volume on a channel.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110542 65c4cc65-6c06-0410-ace0-fbb531ad65f3

is subject to change while we work out the remaining issues.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110499 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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r110474 | qwell | 2008-03-21 09:32:52 -0500 (Fri, 21 Mar 2008) | 7 lines

Don't attempt to do optimizations of gsm on mips platforms either.

(closes issue #12270)
Reported by: zandbelt
Patches:
      026-gsm-mips.patch uploaded by zandbelt (license 33)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110444 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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r110395 | russell | 2008-03-20 18:13:56 -0500 (Thu, 20 Mar 2008) | 9 lines

Shorten the ast_waitfor() timeout from 500 ms to 50 ms in the autoservice thread.
This really should not make a difference except in very rare cases.  That case would
be that all of the channels in autoservice are not generating any frames.  In that
case, this change reduces the potential amount of time that a thread waits in
ast_autoservice_stop() for the autoservice thread to wrap back around to the beginning
of its loop.

(closes issue #12266, reported by dimas)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110396 65c4cc65-6c06-0410-ace0-fbb531ad65f3

problems, but Qwell noticed the typo here.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110339 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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r110336 | russell | 2008-03-20 16:54:58 -0500 (Thu, 20 Mar 2008) | 14 lines

Merged revisions 110335 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r110335 | russell | 2008-03-20 16:53:27 -0500 (Thu, 20 Mar 2008) | 6 lines

Fix some very broken code that was introduced in 1.2.26 as a part of the security
fix.  The dnsmgr is not appropriate here.  The dnsmgr takes a pointer to an address
structure that a background thread continuously updates.  However, in these cases,
a stack variable was passed.  That means that the dnsmgr thread would be continuously
writing to bogus memory.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110337 65c4cc65-6c06-0410-ace0-fbb531ad65f3

other than 8 kHz.  The issue here is that format modules give a "whennext" sample
value, which is used to calculate when to set a timer for to retrieve the next
frame.  However, the zaptel timer operates on 8 kHz samples, so this must be taken
into account.

(another part of issue #12164, reported by milazzo and jsmith, patch by me)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110303 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110272 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110270 65c4cc65-6c06-0410-ace0-fbb531ad65f3

G.722 music on hold working for me.

(issue #12164, reported by milazzo and jsmith, patches by me)

res/res_musiconhold.c:
 - I moved a single line so that the sample queue update happened before
   ast_write().  The reason that this was a bug is that the G.722 frame
   originally says it has 320 samples in it (which is correct).  However,
   when the frame is written to a channel that uses RTP, main/rtp.c modifies
   the frame to cut the number of samples in half before it sends it on
   the wire.  This is to account for the stupid incorrect G.722 spec that
   makes it so we have to lie about the number of samples with RTP.  I should
   probably go and re-work the RTP code so it doesn't modify the frame so
   that a bug like this won't happen in the future.  However, this change to
   MOH is harmless.

main/channel.c:
 - I made two fixes in regards to generator timing.  Generators use samples
   for timing.  However, this code assumed 8 kHz samples.  In one case, it was
   a hard coded 160 samples, that is now written as the sample rate / 50.  The
   other place was dealing with timing a generator based on frames coming from
   the other direction.  However, that would have only worked if the sample
   rates for the formats in both directions were the same.  The code now takes
   into account that the sample rates may differ, and scales the generator
   samples accordingly.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110268 65c4cc65-6c06-0410-ace0-fbb531ad65f3

	1) Number of digits to enter can now be configured
	2) The digits can now match on both first AND last name, instead of only one or the other
(Closes issue #7151)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110237 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110211 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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r110163 | russell | 2008-03-19 17:57:59 -0500 (Wed, 19 Mar 2008) | 5 lines

Fix a bug where when calls on the trunk side hang up while on hold, the state
is not properly reflected.

(closes issue #11990, reported by anakaoka, patched by me)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110164 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This was very misleading.

Early cleanup for issue #11968


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110161 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Rename very poorly named function to reflect what it actually does.  This was causing quite a bit of confusion for me...

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110132 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change adds DNS manager support for registrations not referencing a peer entry. It looks like there is support for DNS manager for realtime peers as well, however it is not implemented correctly. The improper usage occurs when ast_dnsmgr_lookup is called with one of the arguments being an address from the stack to be continually updated. The variable from the stack will go out of scope and dnsmgr will continue to try and update the memory there, causing possible stack corruption. This problem will be worked on next as well as adding DNS manager support for peer entries.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110087 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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r110083 | mmichelson | 2008-03-19 15:33:03 -0500 (Wed, 19 Mar 2008) | 4 lines

Add a missing unlock in the case that memory allocation fails in app_chanspy.
Thanks to Russell for confirming that this was an issue.


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

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r110035 | file | 2008-03-19 16:11:33 -0300 (Wed, 19 Mar 2008) | 4 lines

Add sanity checking for position resuming. We *have* to make sure that the position does not exceed the total number of files present, and we have to make sure that the position's filename is the same as previous. These values can change if a music class is reloaded and give unpredictable behavior.
(closes issue #11663)
Reported by: junky

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110023 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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

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r110019 | file | 2008-03-19 15:20:28 -0300 (Wed, 19 Mar 2008) | 6 lines

Make sure that the mark bit does not incorrectly cause video frame timestamps to be calculated as if they are audio frames.
(closes issue #11429)
Reported by: sperreault
Patches:
      11429-frametype.diff uploaded by qwell (license 4)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110020 65c4cc65-6c06-0410-ace0-fbb531ad65f3