remove gentone and all the headers containing tones that are no longer needed


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96270 65c4cc65-6c06-0410-ace0-fbb531ad65f3

remove background thread and all sound generation mechanisms, as the built-in indications can handle everything that is needed


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

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r96198 | crichter | 2008-01-03 13:08:40 +0100 (Do, 03 Jan 2008) | 1 line

when overlapdial was used and no number was dialed, the call was dropped, now we just jump into the s extension, which makes a lot more sense.
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Feature proposed and patched by: moy
(Closes issue #11282)


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Report and fix by: mvanbaak
(Closes issue #11669)


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r96102 | mmichelson | 2008-01-02 17:46:02 -0600 (Wed, 02 Jan 2008) | 4 lines

We need to reset the membername to NULL on each iteration of this loop, otherwise the result is that
multiple members can have the same name, since the variable was not reset on each iteration of the loop.


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of a hack.  It just asks the core to generate the same tone that it would when
you hear ringback when making an outbound call.  But hey, it works, and you get
the localized ring tone for the appropriate language set on the channel.


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... oops


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Don't use AST_C_DEFINE_CHECK for the two pthread things that may not actually be definitions, they could be enums for example.


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r96024 | russell | 2008-01-02 16:14:28 -0600 (Wed, 02 Jan 2008) | 2 lines

Convert locks of the contexts list in pbx_config to the appropriate rdlock or wrlock

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r96022 | russell | 2008-01-02 16:04:47 -0600 (Wed, 02 Jan 2008) | 2 lines

pbx_dundi only needs a rdlock on the contexts list.

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r96020 | russell | 2008-01-02 16:00:21 -0600 (Wed, 02 Jan 2008) | 2 lines

app_macro only needs a rdlock on the contexts list.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96021 65c4cc65-6c06-0410-ace0-fbb531ad65f3

another checkpoint... chan_zap can now use the new ZT_ECHOCAN_PARAMS ioctl if it is present, but doesn't parse any supplied parameters yet

(this implementation is not very memory efficient as the parameters and their values will be duplicated for each channel that has the same settings, but we can worry about that later once it is working)


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(closes issue #11668, reported and patched by mvanbaak)



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

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r95946 | file | 2008-01-02 16:24:09 -0400 (Wed, 02 Jan 2008) | 4 lines

Allocate a SIP refer structure when performing a transfer using BYE with Also so that the transfer information is properly stored. (AST-2008-001)
(closes issue #11637)
Reported by: greyvoip

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to use AST_STANDARD_APP_ARGS instead of AST_NONSTANDARD_APP_ARGS for determining
member data.



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not '|'.



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improve AC_C_DEFINE_CHECK to not try to evaluate the macro being checked for, but just check for its existence
finish implementation of check for Zaptel HWGAIN support
add check for Zaptel ECHOCANCEL_PARAMS support


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and now just to keep the libresample party going... if the functions from libresample are going to be in the main Asterisk binary, it makes sense for the header that defines them to be available without any special CFLAGS and to out-of-tree modules building against /usr/include/asterisk


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umm... this did not compile on x86-64, and could not possibly have worked on any platform as it was passing string pointers to a function expecting ints


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

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r95890 | mmichelson | 2008-01-02 11:51:22 -0600 (Wed, 02 Jan 2008) | 9 lines

A change to improve the accuracy of queue logging in the case where a member does not
answer during the specified timeout period. Prior to this change, there was a small chance
that the member name recorded in this case would be blank. Also prior to this change, if using
the ringall strategy, if no one answered the call during the specified timeout, the member name
listed in the queue log would randomly be one of the members that was rung.

(closes issue #11498, reported and tested by hloubser, patched by me)


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95891 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Closes issue #11666, patch by Laureano.


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not work for everyone, but it did for some.  This set of changes makes trunk
start again for those having problems.  Instead of building libresample as a
static library, it just links the object files in directly with the asterisk
binary.


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actually check for a function present in libiconv (don't know how this test could have worked before) and don't do the check on Linux/GNU systems because libiconv is not present there and attempting to link with '-liconv' always fails (it's not necessary as the iconv functionality is always available)


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go back to including libresample in the main Asterisk binary, but this time including a small hack to ensure that it does get linked in (and also modify the strip_nonapi script to leave the resample_<foo> symbols alone)


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When the XMPP over TLS/SSL connection resets for some reason, it is
wrongly believed as being secured, which makes the re-connection
process endlessly fail. This was reported by mvanbaak in issue #11644.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95794 65c4cc65-6c06-0410-ace0-fbb531ad65f3

gdb dying while debugging asterisk. The problem seems to be related
with a race in the handling of module_list, which in turn is triggeded
by calling dlopen() on a system which uses initializers to create
locks.



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which make maintaining this file very error prone.

This commit merges the embedded and !embedded versions,
and fixes the C++ version. Eventually we should move to
a single version of the macro.

Too bad C++ doesn't like the C-style struct initializers
    .foo = some_value



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res_resample, and mark codec_resample as dependent upon res_resample.  This
prevents the linker from optimizing away libresample, and also makes it so the
libresample code isn't linked in to multiple places.  (I have another module
in a branch that needs it, too.)


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