https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96103 f38db490-d61c-443f-a65b-d21fe96a405b
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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96079 f38db490-d61c-443f-a65b-d21fe96a405b
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96025 f38db490-d61c-443f-a65b-d21fe96a405b
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96023 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@96021 f38db490-d61c-443f-a65b-d21fe96a405b
(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)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96019 f38db490-d61c-443f-a65b-d21fe96a405b
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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95947 f38db490-d61c-443f-a65b-d21fe96a405b
to use AST_STANDARD_APP_ARGS instead of AST_NONSTANDARD_APP_ARGS for determining
member data.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95945 f38db490-d61c-443f-a65b-d21fe96a405b
finish implementation of check for Zaptel HWGAIN support
add check for Zaptel ECHOCANCEL_PARAMS support
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95937 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@95891 f38db490-d61c-443f-a65b-d21fe96a405b
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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95864 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@95794 f38db490-d61c-443f-a65b-d21fe96a405b
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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95772 f38db490-d61c-443f-a65b-d21fe96a405b
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
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95771 f38db490-d61c-443f-a65b-d21fe96a405b
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.)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95697 f38db490-d61c-443f-a65b-d21fe96a405b
(and probably other systems as well).
Both need libresample.a to be specified in the linking phase,
and cygwin needs <float.h> as other BSD.
The checks for OS-specific headers should really be moved to some
common header though.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95625 f38db490-d61c-443f-a65b-d21fe96a405b
iconv dependency for func_iconv.
This fixes some build issues on CYGWIN and FreeBSD and probably
other platforms where libiconv is not there by default
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95624 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r95577 | mmichelson | 2007-12-31 17:43:13 -0600 (Mon, 31 Dec 2007) | 9 lines
Avoiding a potentially bad locking situation. ast_merge_contexts_and_delete writelocks the conlock, then
calls ast_hint_extension, which attempts to readlock the same lock. Recursion with read-write locks is
dangerous, so the inner lock needs to be removed. I did this by copying the "guts" of ast_hint_extension
into ast_merge_contexts_and_delete (sans the extra lock).
(this change is inspired by the locking problems seen in issue #11080, but I have no idea if this is the
problematic area experienced by the reporters of that issue)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95578 f38db490-d61c-443f-a65b-d21fe96a405b
to an 8 kHz endpoint, then codec_resample will automatically be used to properly
resample the audio before sending it to/from chan_console.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95527 f38db490-d61c-443f-a65b-d21fe96a405b
This commit imports libresample for use in Asterisk. It also adds a new codec
module, codec_resample. This module uses libresample to re-sample signed linear
audio between 8 kHz and 16 kHz.
It also provides an alternative for converting between 16 kHz G.722 and 8 kHz
signed linear when using G.722, which will likely be useful as some people have
complained about volume issues when the current codec_g722 converts to 8 kHz
signed linear. But, to test this, you will have to disable the g722-to-slin and
g722-to-slin16 translators in codec_g722.c.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95501 f38db490-d61c-443f-a65b-d21fe96a405b
