Luigi Rizzo
b90f91c3b4
do not dereference p if we know it is NULL.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44745 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
1b64c41369
Fix some debug output for setsockopt for TOS
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
259f0d76b6
- move definition of global_autoframing to the same place as other globals
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- set initial value at load/reload
- Add questionmarks for someone to fill in for doxygen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44717 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
12ef70f0c0
Add/change doxygen and comments
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19 years ago
Olle Johansson
9cc9c38250
Explain usage of DEFAULT_SIP_PORT
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44706 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
64aae4a887
Do *NOT* use DEFAULT_SIP_PORT in these comparisions, since users may change
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that, but the protocol clearly states that if we DO NOT mention a port it
is 5060. DEFAULT_SIP_PORT is whatever we default to listen to.
I believe it's the third time I revert a patch like this.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44697 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Paul Cadach
1ed8fb571f
Merged revisions 44684 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r44684 | pcadach | 2006-10-07 20:39:34 +0600 (Сбт, 07 Окт 2006) | 1 line
Propagate caller's transfer capability too
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19 years ago
Luigi Rizzo
901a66c401
slightly restructure the code that computes the channel's name
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19 years ago
Luigi Rizzo
a1a5824869
put repeated code to set nat mode in a function.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44674 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Luigi Rizzo
bc1e5f77af
put common code in a function to avoid repetitions.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44664 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Luigi Rizzo
b37dc86c74
remove hardwired usage of 5060, use DEFAULT_SIP_PORT instead
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44663 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Luigi Rizzo
e691246b6e
improve and document function get_in_brackets(), introducing
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a helper function find_closing_quote() of more general use.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44651 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Luigi Rizzo
6466c2ebf9
when possible, use ast_set2_flags instead of ast_set/ast_clr .
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Also mark XXX some dubious places.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44641 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Luigi Rizzo
ccca5843fd
Two things:
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1. slightly rearrange/simplify the parsing of the argument in sip_register.
This brings in a patch that has been in Mantis (5834) for ages,
and is the larger part of the commit;
2. implement the "contact" option for peers, similar to the one in users.conf:
If you put a "contact" option with a non-empty argument (e.g. contact=123)
in a peer section, asterisk will register with the provider as if you had a
register= username:secret@host/contact
line in the general section.
The latter is a very small is a new feature so i am not putting it
in the 1.4 branch, although the "contact" option in user.conf is
already in the 1.4 branch and so it wouldn't be too strange to
merge it.
Note that the implementation of "contact" is much simpler than
the one in 5834, and limited to a few lines in build_peer().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44566 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Olle Johansson
cb127f4baa
Remove deprecated "incominglimit" config option
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44545 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
0b26b56316
Merged revisions 44486 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r44486 | file | 2006-10-05 14:01:51 -0400 (Thu, 05 Oct 2006) | 2 lines
One more T.38 fix! Don't leave a reinvite hanging by a thread if the other side is already setup with T.38
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19 years ago
Joshua Colp
9924c3e2e9
Merged revisions 44450 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r44450 | file | 2006-10-04 22:40:40 -0400 (Wed, 04 Oct 2006) | 2 lines
Don't totally bail out if T.38 was negotiated
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19 years ago
Kevin P. Fleming
975ef9de9d
Merged revisions 44433 via svnmerge from
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r44433 | kpfleming | 2006-10-04 20:30:05 -0500 (Wed, 04 Oct 2006) | 10 lines
Merged revisions 44432 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r44432 | kpfleming | 2006-10-04 20:27:57 -0500 (Wed, 04 Oct 2006) | 2 lines
fix Polycom presence notification again
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19 years ago
Kevin P. Fleming
50bcacbdbd
Merged revisions 44393 via svnmerge from
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r44393 | kpfleming | 2006-10-04 16:17:30 -0500 (Wed, 04 Oct 2006) | 11 lines
Merged revisions 44392 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r44392 | kpfleming | 2006-10-04 16:15:29 -0500 (Wed, 04 Oct 2006) | 3 lines
remove workaround for old Polycom firmware SUBSCRIBE requests
add workaround for new Polycom firmware SUBSCRIBE requests (bug is known to exist in 2.0.1 firmware)
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19 years ago
Kevin P. Fleming
3c876af5cf
Merged revisions 44378 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r44378 | kpfleming | 2006-10-04 14:47:22 -0500 (Wed, 04 Oct 2006) | 4 lines
update thread creation code a bit
reduce standard thread stack size slightly to allow the pthreads library to allocate the stack+data and not overflow a power-of-2 allocation in the kernel and waste memory/address space
add a new stack size for 'background' threads (those that don't handle PBX calls) when LOW_MEMORY is defined
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19 years ago
Matt O'Gorman
7294ba3852
Merged revisions 44312 via svnmerge from
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https://svn.digium.com/svn/asterisk/branches/1.4
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r44312 | mogorman | 2006-10-03 17:35:43 -0500 (Tue, 03 Oct 2006) | 2 lines
fix issue with dialing client without resource.
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19 years ago
Paul Cadach
eb7f97f2d6
Merged revisions 44283,44286 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r44283 | pcadach | 2006-10-04 00:30:48 +0600 (Срд, 04 Окт 2006) | 1 line
Fix preparation of type and presentation of calling number
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r44286 | pcadach | 2006-10-04 00:42:20 +0600 (Срд, 04 Окт 2006) | 1 line
Change default presentation indicator to "user provided not screened" if octet 3a missed in CallingPartyNumber IE
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19 years ago
Joshua Colp
6727e8f599
Merged revisions 44284 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r44284 | file | 2006-10-03 14:35:55 -0400 (Tue, 03 Oct 2006) | 2 lines
Use VideoSupport instead so it is considered a valid XML attribute name. (issue #8075 reported by renemendoza)
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19 years ago
Matt O'Gorman
ae8cc3e18b
bug #8076 check option_debug before printing to debug channel.
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patch provided in bugnote, with minor changes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44253 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Joshua Colp
fadd9e7831
Merged revisions 44215 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r44215 | file | 2006-10-02 16:11:02 -0400 (Mon, 02 Oct 2006) | 10 lines
Merged revisions 44213 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r44213 | file | 2006-10-02 16:07:59 -0400 (Mon, 02 Oct 2006) | 2 lines
Change the fd on the I/O context in case it changed during the reload, which is indeed possible. (issue #7943 reported by eclubb)
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19 years ago
Paul Cadach
b018f59307
Merged revisions 44166 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r44166 | pcadach | 2006-10-02 23:15:11 +0600 (Пнд, 02 Окт 2006) | 1 line
Optimization of oh323_indicate(): less locks - less problems, plus single exit point
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19 years ago
Mark Spencer
847fe77eb9
Don't use channel when you don't mean a channel
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19 years ago
Paul Cadach
34258cf202
Merged revisions 44135 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r44135 | pcadach | 2006-10-02 01:32:24 +0600 (Пнд, 02 Окт 2006) | 1 line
Do not simulate any audio tones if we got PROGRESS message
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44136 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Paul Cadach
d2087e6a09
Merged revisions 44078 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r44078 | pcadach | 2006-09-30 22:12:23 +0600 (Сбт, 30 Сен 2006) | 6 lines
Fix issue #7928 correctly. Next is a comment of previous fix:
Issue #7928 - Don't send both 404 and 503. Fix by phsultan with
a small fix by me, myself or I. Thanks, Philippe!
(This was caused by my changes to the transaction handling)
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19 years ago
Paul Cadach
031cd452d9
Merged revisions 44068 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r44068 | pcadach | 2006-09-30 10:37:39 +0600 (Сбт, 30 Сен 2006) | 14 lines
Found some buggy SIP clients (phones Planet VIP-153T firmware
1.0, Linksys PAP2 firmware 3.1.9(LSc)) which sends ACK not on OK
message only (when remote party answers) but on RINGING message
too, so when we send 200 OK message, we get unidentified ACK
message (because INVITE acknowledged on RINGING message already),
so 200 OK retransmits within its retransmission interval then
call gets dropped.
If someone else knows how to provide workaround for such cases,
please, fix it in correct way.
Thanks to ssh from #asteriskru for provide access to his box to
study and fix this case.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44069 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
Kevin P. Fleming
e92c34cc38
Merged revisions 44055 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r44055 | kpfleming | 2006-09-29 17:47:40 -0500 (Fri, 29 Sep 2006) | 2 lines
fix a few build system bugs, and convert Makefiles to be compatible with GNU make 3.80
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19 years ago
Paul Cadach
6c273d37cc
Merged revisions 44034,44042-44043 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r44034 | pcadach | 2006-09-30 02:43:13 +0600 (Сбт, 30 Сен 2006) | 1 line
Fake display name by called number on incoming calls (until passing connected number/connected name is not implemented)
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r44042 | pcadach | 2006-09-30 03:05:43 +0600 (Сбт, 30 Сен 2006) | 1 line
Set TON/PRESENTATION information more carefully when no CallingNumber IE available
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r44043 | pcadach | 2006-09-30 03:09:10 +0600 (Сбт, 30 Сен 2006) | 1 line
Compile first, please
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19 years ago
Paul Cadach
925708f307
Merged revisions 44022 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r44022 | pcadach | 2006-09-30 01:06:55 +0600 (Сбт, 30 Сен 2006) | 3 lines
Properly pass TON/PRESENTATION information - original
H323Connection::SendSignalSetup() destroys Q.931 fields.
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19 years ago
Kevin P. Fleming
89050169c8
Merged revisions 43996-43997,44008,44011-44012 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43996 | kpfleming | 2006-09-29 11:47:05 -0500 (Fri, 29 Sep 2006) | 2 lines
another cross-compile fix
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r43997 | kpfleming | 2006-09-29 11:52:27 -0500 (Fri, 29 Sep 2006) | 2 lines
support --without-curl in configure script
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r44008 | kpfleming | 2006-09-29 13:25:49 -0500 (Fri, 29 Sep 2006) | 2 lines
don't abuse CFLAGS and LDFLAGS for build of Asterisk components, because they are also then used for non-Asterisk components (like menuselect); use our own variables instead
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r44011 | kpfleming | 2006-09-29 13:40:17 -0500 (Fri, 29 Sep 2006) | 2 lines
missed one conversion to ASTCFLAGS
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r44012 | kpfleming | 2006-09-29 13:49:07 -0500 (Fri, 29 Sep 2006) | 2 lines
yet another place where we were not using the correct CFLAGS by default
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19 years ago
Paul Cadach
ee4141c375
Merged revisions 44009 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r44009 | pcadach | 2006-09-30 00:30:34 +0600 (Сбт, 30 Сен 2006) | 1 line
Pass TON/PRESENTATION information too
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19 years ago
Joshua Colp
6c0391c42a
One must remember to unlock their list... thanks to Qwell for letting me into his box
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19 years ago
Joshua Colp
29d27bfe0d
Merged revisions 43933 via svnmerge from
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r43933 | file | 2006-09-28 14:05:43 -0400 (Thu, 28 Sep 2006) | 2 lines
Put in missing \ns on the end of ast_logs (issue #7936 reported by wojtekka)
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19 years ago
Paul Cadach
134a9ac96d
Merged revisions 43918 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43918 | pcadach | 2006-09-28 23:34:19 +0600 (Чтв, 28 Сен 2006) | 1 line
Extend call establishment timeout
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19 years ago
Joshua Colp
773e105d91
Merged revisions 43915 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43915 | file | 2006-09-28 13:31:09 -0400 (Thu, 28 Sep 2006) | 2 lines
Make sure the pvt exists before accessing it again as it may have gone away (issue #7562 reported by Seb7 and issue #7939 reported by sorg)
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19 years ago
Joshua Colp
8785d7229f
Add jitterbuffer information to sip list settings (issue #7945 reported by sergee)
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19 years ago
Paul Cadach
5bfa99306f
Merged revisions 43877 via svnmerge from
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r43877 | pcadach | 2006-09-28 21:56:21 +0600 (Чтв, 28 Сен 2006) | 1 line
Fix compiler warning
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19 years ago
Paul Cadach
2719c20e4e
Merged revisions 43861-43862 via svnmerge from
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r43861 | pcadach | 2006-09-28 18:47:23 +0600 (Чтв, 28 Сен 2006) | 1 line
Put attribute tag at correct place
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r43862 | pcadach | 2006-09-28 18:58:22 +0600 (Чтв, 28 Сен 2006) | 1 line
Force remote side to start media on outgoing PROGRESS message
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19 years ago
Christian Richter
6001354638
Merged revisions 43852 via svnmerge from
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r43852 | crichter | 2006-09-28 13:03:05 +0200 (Do, 28 Sep 2006) | 9 lines
Merged revisions 43764 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r43764 | crichter | 2006-09-27 14:51:03 +0200 (Mi, 27 Sep 2006) | 1 line
fixed a bug which led to chan_list zombies, when the call could not be properly established in misdn_call. also removed the ACK_HDLC stuff which is not really needed.
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19 years ago
Christian Richter
edb0abd57d
Merged revisions 43775 via svnmerge from
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r43775 | crichter | 2006-09-27 18:24:51 +0200 (Mi, 27 Sep 2006) | 1 line
removed the chan_misdn versioning, since asterisk has it's own
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19 years ago
Paul Cadach
47a0e0807c
Merged revisions 43635,43843-43844,43846 via svnmerge from
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r43635 | pcadach | 2006-09-26 03:26:12 +0600 (Втр, 26 Сен 2006) | 1 line
Fix ASN1 description of non-standard Cisco extensions
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r43843 | pcadach | 2006-09-28 12:01:37 +0600 (Чтв, 28 Сен 2006) | 1 line
Don't treat unknown control frames as voice
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r43844 | pcadach | 2006-09-28 12:02:45 +0600 (Чтв, 28 Сен 2006) | 1 line
Don't warn on HOLD/UNHOLD control frames
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r43846 | pcadach | 2006-09-28 16:51:21 +0600 (Чтв, 28 Сен 2006) | 1 line
Do not open transmit channel until TCS is received
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19 years ago
Paul Cadach
9cf1f14ed5
Handle HOLD/RETRIEVE notifications
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19 years ago
Joshua Colp
dad32d9d71
I *think* this is the last list in chan_iax2
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19 years ago
Joshua Colp
a5770f4e72
Merged revisions 43783 via svnmerge from
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r43783 | file | 2006-09-27 13:00:31 -0400 (Wed, 27 Sep 2006) | 2 lines
Get rid of two functions from a time now past (we THINK these are from pre-recursive lock time) that may be contributing to two open issues on the bug tracker (7562/7939) and that has the potential to just make bad things happen if the timing is right.
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19 years ago
Matthew Fredrickson
6b3a6a1104
Fix some little things
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19 years ago
Matthew Fredrickson
e5435429de
Add CLI block and unblock circuit commands for SS7.
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19 years ago
Joshua Colp
fcad797dfd
Merged revisions 43774 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43774 | file | 2006-09-27 12:23:12 -0400 (Wed, 27 Sep 2006) | 2 lines
Make rfc2833compensate a global option.
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19 years ago
Paul Cadach
e1e63b2355
Use ast_strdupa() instead of strdup(), thanks to sergee
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19 years ago
Joshua Colp
481265e347
Clean up code and convert last two things (firmware/dialplan cache) to linked list macros.
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19 years ago
Joshua Colp
98873d34bd
Merged revisions 43707 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43707 | file | 2006-09-26 16:47:26 -0400 (Tue, 26 Sep 2006) | 10 lines
Merged revisions 43705 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r43705 | file | 2006-09-26 16:38:06 -0400 (Tue, 26 Sep 2006) | 2 lines
Use proper type to represent the group variable (issue #8025 reported by makoto)
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19 years ago
Joshua Colp
36a9656234
Merged revisions 43697 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43697 | file | 2006-09-26 16:19:33 -0400 (Tue, 26 Sep 2006) | 2 lines
Strip options off the argument passed for devicestate in chan_local. (issue #8034 reported by pcardozo)
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19 years ago
Jason Parker
8b42bb0f3a
Merged revisions 43650 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43650 | qwell | 2006-09-26 08:33:47 -0700 (Tue, 26 Sep 2006) | 11 lines
Add proper codec support to chan_skinny. Works with at least ulaw, alaw, and g729a.
This is technically a "new feature", but there are justifications for it.
I found a bug with the recent rtp packetization changes, which caused the media setup to
fail under certain circumstances, particularly when using allow=all, or having no allow=
statements (globally or on the device).
I could have either removed the rtp packetization features, or I could add proper codec
support (which, without, I think most people would consider to be a bug anyways).
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19 years ago
Paul Cadach
c479c25182
Support for negotiation and receiption of Cisco's RTP DTMF
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19 years ago
Paul Cadach
66e1d206f5
Disable fastStart if requested by remote side
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19 years ago
Paul Cadach
1af96a0b21
Specify RFC2833 payload on dtmfmode option rather than dtmfcodec option (deprecated)
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19 years ago
Paul Cadach
d62390a65c
DTMF mode is bitmask, not valued field
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19 years ago
Paul Cadach
aec46b99f5
Define Cisco RTP capability
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19 years ago
Paul Cadach
ecdfdb6ecb
Specify non-standard data independedly on OpenH323's codec name (it can be easily changed)
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19 years ago
Paul Cadach
258f1082ec
Define DTMF payload types
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19 years ago
Russell Bryant
2fa3574c57
Merged revisions 43564 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43564 | russell | 2006-09-24 10:58:10 -0400 (Sun, 24 Sep 2006) | 5 lines
Fix a CLI command registration issue where an erroneous message claiming that
"iax2 show provisioning" was already registered. This was because this command
was registering itself as both the command, as well as the command it is
deprecating. (issue #8022 , reported by bjweeks, fixed by myself)
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19 years ago
Russell Bryant
c8b90b75c9
Merged revisions 43553 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43553 | russell | 2006-09-24 09:53:35 -0400 (Sun, 24 Sep 2006) | 12 lines
Merged revisions 43552 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r43552 | russell | 2006-09-24 09:50:30 -0400 (Sun, 24 Sep 2006) | 4 lines
Check to see if the channel that is activating the IAXPEER function is actually
an IAX2 channel before proceeding to process it to avoid crashing.
(issue #8017 , reported by admott, fixed by myself)
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19 years ago
Paul Cadach
56fb349222
Avoid possible deadlock on channel destruction
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19 years ago
Jason Parker
cb8065e5c3
Merged revisions 43518 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43518 | qwell | 2006-09-22 15:12:12 -0700 (Fri, 22 Sep 2006) | 4 lines
Allow chan_skinny.so to be unloaded properly.
Remove reload support, since it doesn't actually...work.
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19 years ago
Paul Cadach
23db39367c
Move from h.323 to h323 command prefix
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19 years ago
Paul Cadach
2ccf0b434a
Fix compilation warnings
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19 years ago
Paul Cadach
747f091a77
Use own factory for our OpalMediaFormats too
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19 years ago
Paul Cadach
50ed891da2
Fix our capability's factory
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19 years ago
Russell Bryant
34da4819c1
Merged revisions 43482 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43482 | russell | 2006-09-22 11:42:44 -0400 (Fri, 22 Sep 2006) | 3 lines
return AST_MODULE_LOAD_DECLIDE if mISDN could not be configured
(issue #8006 , Mithraen)
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19 years ago
Luigi Rizzo
4d188126d9
style fix:
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move variable declaration at the beginning of the block.
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19 years ago
Paul Cadach
ccdbf69318
Add missing include
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19 years ago
Jason Parker
81bfa44bb8
Merged revisions 43469 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43469 | qwell | 2006-09-21 20:01:16 -0700 (Thu, 21 Sep 2006) | 4 lines
First shot at unload_module in chan_skinny..
More to come.
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19 years ago
Matt O'Gorman
08368f00ab
Merged revisions 43466 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43466 | mogorman | 2006-09-21 18:50:56 -0500 (Thu, 21 Sep 2006) | 2 lines
updates for better compontent support
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19 years ago
Joshua Colp
323527b4f3
Oh look more changes, but these are my own! (Clean up module load functions)
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19 years ago
Jason Parker
57504cf0f1
Suppress compiler warnings
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19 years ago
Joshua Colp
f2e34318f8
Clean up chan_alsa load module function (issue #8000 reported by Mithraen)
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19 years ago
Tilghman Lesher
f14dc5fab4
And some deprecated APIs and modifications to documentation
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19 years ago
Joshua Colp
794d34f90c
Merged revisions 43456 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43456 | file | 2006-09-21 18:21:40 -0400 (Thu, 21 Sep 2006) | 2 lines
Some more clean up in the load function for chan_oss (issue #8002 reported by Mithraen with minor mods by moi)
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19 years ago
Joshua Colp
a5e4e31654
Merged revisions 43454 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43454 | file | 2006-09-21 18:12:09 -0400 (Thu, 21 Sep 2006) | 2 lines
Clean up chan_mgcp's module load function (issue #8001 reported by Mithraen with mods by moi)
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19 years ago
Tilghman Lesher
6ae8741518
Lots more removal of deprecated things
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19 years ago
Joshua Colp
1c764935f2
SS7 marked the start of an open season for trunk again but here's something minor - abstract early bridging into the technology so that we don't always assume they use RTP and try it.
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19 years ago
Paul Cadach
d2b0446c90
Declare our own media formats to not rely on OpenH323 configuration
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19 years ago
Paul Cadach
5acd11f180
Introduce Cisco G.726-32 capability (g726aal2 form)
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19 years ago
Matthew Fredrickson
bd76cda68d
Merge in SS7 changes.... need to still cleanup zapata.conf
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19 years ago
Mark Spencer
c58ae729cd
Fix agent parsing of users.conf
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19 years ago
Jason Parker
0cfaffac96
Fix some chan_zap warning messages, when HAVE_PRI is not defined.
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19 years ago
Tilghman Lesher
d70fe791c2
A few misses from constification
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19 years ago
Tilghman Lesher
70af28270d
Constify the result of a config retrieval function, to avoid mutilation (issue 7983).
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19 years ago
Paul Cadach
e57d521462
Split capability definitions into own source file
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19 years ago
Paul Cadach
aa92ebffea
Remove unnecessary (long time ago commented out) code
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19 years ago
Paul Cadach
a704e298e7
Fix missed extension replacement
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19 years ago
Paul Cadach
84946195e9
Change file naming to be conform OpenH323 rules
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19 years ago
Joshua Colp
86421e3a7d
Do clean for h323 directory too
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19 years ago
Joshua Colp
1ce4a427d3
Merge in latest round of chan_h323 changes. These are all isolated to chan_h323 so meh.
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19 years ago
Kevin P. Fleming
0964265657
Merged revisions 43314 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r43314 | kpfleming | 2006-09-20 00:08:05 -0500 (Wed, 20 Sep 2006) | 2 lines
make some more functions static
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19 years ago
BJ Weschke
e8606d9c57
Merged revisions 42133 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r42133 | bweschke | 2006-09-06 14:16:41 -0400 (Wed, 06 Sep 2006) | 6 lines
Look ma! No more deadlocks! <sic>
As posted from #7458 and others similar to it in Mantis:
p->app_lock was a mutex really designed for use with agents not in callback mode. That being the case, I've tried to code it so that when callback mode is used, the app_lock mutex will not be locked/unlocked at all. Please let me know how you make out - and if you continue to deadlock now, please reproduce the deadlock logging information and post to Mantis.
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19 years ago
Kevin P. Fleming
1485fa4c24
move this header to include where it belongs
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19 years ago