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

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r47333 | kpfleming | 2006-11-08 12:07:16 -0600 (Wed, 08 Nov 2006) | 2 lines

add simple fix for SDP to report proper sample rate for G.722 media sessions

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

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r47331 | russell | 2006-11-08 12:03:09 -0500 (Wed, 08 Nov 2006) | 5 lines

I occasionally get email from users that are trying to figure out what this
does, or due to some misunderstanding as to what it is supposed to do, can't
get it to work.  So, I have added some text here to hopefully explain what
this application does and does not do.

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r47329 | russell | 2006-11-08 11:55:35 -0500 (Wed, 08 Nov 2006) | 2 lines

Make this module build again

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

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r47327 | russell | 2006-11-08 11:31:59 -0500 (Wed, 08 Nov 2006) | 4 lines

Copy the macros from libtool.m4 to our own acinclude.m4 such that libtool is
no longer required to be installed to be able to generate the configure
script.

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r47323 | russell | 2006-11-08 11:08:50 -0500 (Wed, 08 Nov 2006) | 3 lines

Remove aclocal.m4 from the tree since it is just an intermediate file created
while generating the configure script.

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avoid doing p > 0 when p is a pointer;
move a lock closer to the place where it is needed

Approved By: oej



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Same as for peers and users, replace ASTOBJ_UNREF(r, sip_registry_destroy)
with unref_registry(r);

Approved By: oej



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Replace ASTOBJ_UNREF(peer, sip_destroy_peer) with unref_peer(peer);

This places the name of the destructor in one place only (where it
should be), eliminates the chance of errors in case you specify the wrong
destructor, and also lets the compiler do type checking on the argument,
again helping with keeping the code clean.

Same for users.
remove two duplicate definitions.

Approved By: oej



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47316 65c4cc65-6c06-0410-ace0-fbb531ad65f3

hide dialoglist lock/unlocking in wrapper functions.

Approved By: oej



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47315 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The compiler is wrong, but it has the last word.



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Overall i think the previous change to ast_channel_alloc()
to close bug 7506 should have been done by defining
an ast_set_callerid_noevent() function that does the
setting without generating the event.
Lot less code duplication, and easier to handle.



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while generating the configure script.


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Needs to go in 1.4 too, but ENOTIME now



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to ease portability problems



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

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r47287 | file | 2006-11-07 15:14:58 -0500 (Tue, 07 Nov 2006) | 2 lines

This is not the commit you are looking for...

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r47284 | file | 2006-11-07 15:08:52 -0500 (Tue, 07 Nov 2006) | 2 lines

Make MOH work as it did before in chan_local, without this then it can go funky when transfers and MOH are involved. (issue #7671 reported by jmls)

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r47279 | kpfleming | 2006-11-07 12:56:21 -0600 (Tue, 07 Nov 2006) | 2 lines

clean up sample config, and make native file playback the more obvious default choice

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also a better implementation of several parts of the original work.
patch provided by 8033 with major upgrades. minor differences from 1.4 patch do to
changes in app_voicemail


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and also that it doesn't really work as claimed.



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

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r47239 | russell | 2006-11-06 20:25:10 -0500 (Mon, 06 Nov 2006) | 13 lines

Merged revisions 47238 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r47238 | russell | 2006-11-06 20:22:58 -0500 (Mon, 06 Nov 2006) | 5 lines

If random order is enabled for files mode music on hold, set a random initial
position, instead of always starting at the first file, and doing the random
operation only when switching to the next file.
(bug reported by John Lange on the asterisk-dev mailing list)

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Add support for manager hooks, so you could fire off manager events over IRC if you were crazy enough. (issue #5161 reported by anthm with mods by moi)


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together and comment them out by default.


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Hmmm. I did not change pbx_dundi and yet it doesn't compile ;-)


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