https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47391 | russell | 2006-11-09 16:26:27 -0500 (Thu, 09 Nov 2006) | 7 lines
Work around an issue that caused menuselect to display a bogus description for
app_voicemail and chan_zap. These modules use some preprocessor directives to
determine what it will report to Asterisk as its description. However, the way
we extract this information from the source files for menuselect is not smart
enough to figure this out.
(issue #8326, #8328)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@47392 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47380 | file | 2006-11-09 11:53:25 -0500 (Thu, 09 Nov 2006) | 10 lines
Merged revisions 47379 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r47379 | file | 2006-11-09 11:48:05 -0500 (Thu, 09 Nov 2006) | 2 lines
Don't include compiler.h on kernels 2.6.18 and higher as, well, it's apparently going to be removed. This should make all you FC6 fans happy as your Asterisk will now build without any mods.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@47382 f38db490-d61c-443f-a65b-d21fe96a405b
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r47369 | russell | 2006-11-09 11:04:38 -0500 (Thu, 09 Nov 2006) | 3 lines
Fix argument parsing for the "core show profile" CLI command
(fixed by rizzo in his branch, team/rizzo/astobj2)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@47371 f38db490-d61c-443f-a65b-d21fe96a405b
- Fix documentation for sip_pvt_lock/unlock - doxygen does not inherit like zapata.conf !!!
- Change doc for a sip_pvt setting
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@47341 f38db490-d61c-443f-a65b-d21fe96a405b
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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@47332 f38db490-d61c-443f-a65b-d21fe96a405b
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@47330 f38db490-d61c-443f-a65b-d21fe96a405b
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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@47328 f38db490-d61c-443f-a65b-d21fe96a405b
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@47325 f38db490-d61c-443f-a65b-d21fe96a405b
avoid doing p > 0 when p is a pointer;
move a lock closer to the place where it is needed
Approved By: oej
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@47318 f38db490-d61c-443f-a65b-d21fe96a405b
Same as for peers and users, replace ASTOBJ_UNREF(r, sip_registry_destroy)
with unref_registry(r);
Approved By: oej
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@47317 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@47316 f38db490-d61c-443f-a65b-d21fe96a405b
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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@47306 f38db490-d61c-443f-a65b-d21fe96a405b
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
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@47278 f38db490-d61c-443f-a65b-d21fe96a405b
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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@47240 f38db490-d61c-443f-a65b-d21fe96a405b
