https://origsvn.digium.com/svn/asterisk/branches/1.4
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r80360 | russell | 2007-08-22 14:53:30 -0500 (Wed, 22 Aug 2007) | 5 lines
Juggie in #asterisk-dev was reporting problems where fgets would return
without reading the whole line when using fastagi. When this happens,
errno was set to EINTR or EAGAIN. This patch accounts for the possibility
and lets fgets continue in that case.
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check_expr binary when building with LOW_MEMORY defined.
(reported by Brian Capouch on the asterisk-dev list, patch by me)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r80257 | russell | 2007-08-22 11:21:58 -0500 (Wed, 22 Aug 2007) | 4 lines
Honor the contents of the COPTS variable as custom target CFLAGS. Apparently
this is what openwrt does.
(reported by Brian Capouch on the asterisk-dev list, patch by me)
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r80166 | murf | 2007-08-21 10:36:34 -0600 (Tue, 21 Aug 2007) | 1 line
This patch solves problem 1 in 8126; it should not slow down the alaw codec, but should prevent signal degradation via multiple trips thru the codec. Fossil estimates the twice thru this codec will prevent fax from working. 4-6 times thru would result hearable, noticeable, voice degradation.
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r80183 | russell | 2007-08-21 13:42:15 -0500 (Tue, 21 Aug 2007) | 7 lines
Don't record SIP dialog history if it's not turned on. Also, put an upper
limit on how many history entires will be stored for each SIP dialog. It is
currently set to 50, but can be increased if deemed necessary.
(closes issue #10421, closes issue #10418, patches suggested by jmoldenhauer,
patches updated by me)
(Security implications documented in AST-2007-020)
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r80086 | mmichelson | 2007-08-20 16:39:17 -0500 (Mon, 20 Aug 2007) | 5 lines
After a discussion on #asterisk-dev, it was decided that this should be in 1.4 as well.
(issue #10424, reported and patched by irroot)
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r79912 | russell | 2007-08-17 16:01:43 -0500 (Fri, 17 Aug 2007) | 4 lines
Avoid a crash in the handling of DTMF based Caller ID. It is valid for
ast_read to return NULL in the case that the channel has been hung up.
(crash reported by anonymouz666 on IRC in #asterisk-dev)
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r79906 | mmichelson | 2007-08-17 14:14:05 -0500 (Fri, 17 Aug 2007) | 6 lines
Patch allows for more seamless transition from file storage voicemail to ODBC storage voicemail.
If a retrieval of a greeting from the database fails, but the file is found on the file system, then
we go ahead an insert the greeting into the database. The result of this is that people who
switch from file storage to ODBC storage do not need to rerecord their voicemail greetings.
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(closes issue #10430)
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r79904 | qwell | 2007-08-17 14:12:19 -0500 (Fri, 17 Aug 2007) | 11 lines
Don't send a semicolon over the wire in sip notify messages.
Caused by fix for issue 9938.
I basically took the code that existed before 9938 was fixed, and
copied it into a new function - ast_unescape_semicolon
There should be very few places this will be needed (pbx_config
does NOT need this (see issue 9938 for details))
Issue 10430, patch by me, with help/ideas from murf (thanks murf).
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in place of a very common construct. I also used it in a number of places
in chan_sip.
if (id > -1)
ast_sched_del(sched, id);
id = ast_sched_add(sched, ...);
changes to:
ast_sched_replace(id, sched, ...);
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r79857 | russell | 2007-08-17 08:37:08 -0500 (Fri, 17 Aug 2007) | 5 lines
Fix some crashes in chan_sip. This patch changes various places that add items
to the scheduler to ensure that they don't overwrite the ID of a previously
scheduled item. If there is one, it should be removed.
(closes issue #10391, closes issue #10256, probably others, patch by me)
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r79833 | crichter | 2007-08-17 10:22:36 +0200 (Fr, 17 Aug 2007) | 1 line
sometimes we don't need to signal dtmf tones to asterisk, we just want them to go through as inband. Otherwise they might be generated by the other channel partner and then there is a double tone.
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