Olle Johansson
9cc9c38250
Explain usage of DEFAULT_SIP_PORT
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2006-10-07 16:03:36 +00:00
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.
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2006-10-07 15:38:18 +00:00
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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2006-10-07 14:48:32 +00:00
Luigi Rizzo
901a66c401
slightly restructure the code that computes the channel's name
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2006-10-07 12:50:42 +00:00
Luigi Rizzo
a1a5824869
put repeated code to set nat mode in a function.
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2006-10-07 12:20:16 +00:00
Luigi Rizzo
bc1e5f77af
put common code in a function to avoid repetitions.
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2006-10-07 11:28:38 +00:00
Luigi Rizzo
b37dc86c74
remove hardwired usage of 5060, use DEFAULT_SIP_PORT instead
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2006-10-07 11:11:39 +00:00
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.
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2006-10-07 09:36:09 +00:00
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.
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2006-10-07 08:56:59 +00:00
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().
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2006-10-06 15:41:12 +00:00
Olle Johansson
cb127f4baa
Remove deprecated "incominglimit" config option
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2006-10-06 09:01:09 +00:00
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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2006-10-05 18:03:34 +00:00
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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2006-10-05 02:43:07 +00:00
Kevin P. Fleming
975ef9de9d
Merged revisions 44433 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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2006-10-05 01:30:44 +00:00
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
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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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2006-10-04 21:20:55 +00:00
Kevin P. Fleming
3c876af5cf
Merged revisions 44378 via svnmerge from
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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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2006-10-04 19:51:38 +00:00
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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2006-10-03 22:36:51 +00:00
Paul Cadach
eb7f97f2d6
Merged revisions 44283,44286 via svnmerge from
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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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2006-10-03 18:47:38 +00:00
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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2006-10-03 18:37:06 +00:00
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.
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2006-10-03 15:53:07 +00:00
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
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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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2006-10-02 20:13:26 +00:00
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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2006-10-02 18:57:49 +00:00
Mark Spencer
847fe77eb9
Don't use channel when you don't mean a channel
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2006-10-02 02:38:52 +00:00
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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2006-10-01 19:40:00 +00:00
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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2006-09-30 16:15:53 +00:00
Paul Cadach
031cd452d9
Merged revisions 44068 via svnmerge from
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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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2006-09-30 04:41:04 +00:00
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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2006-09-29 22:48:43 +00:00
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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2006-09-29 21:13:46 +00:00
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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2006-09-29 19:13:26 +00:00
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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2006-09-29 18:54:21 +00:00
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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2006-09-29 18:35:44 +00:00
Joshua Colp
6c0391c42a
One must remember to unlock their list... thanks to Qwell for letting me into his box
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2006-09-29 01:36:26 +00:00
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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2006-09-28 18:09:01 +00:00
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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2006-09-28 17:36:33 +00:00
Joshua Colp
773e105d91
Merged revisions 43915 via svnmerge from
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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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2006-09-28 17:32:46 +00:00
Joshua Colp
8785d7229f
Add jitterbuffer information to sip list settings (issue #7945 reported by sergee)
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2006-09-28 17:07:37 +00:00
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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2006-09-28 16:01:48 +00:00
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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2006-09-28 13:02:30 +00:00
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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2006-09-28 11:32:32 +00:00
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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2006-09-28 11:27:09 +00:00
Paul Cadach
47a0e0807c
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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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2006-09-28 11:12:58 +00:00
Paul Cadach
9cf1f14ed5
Handle HOLD/RETRIEVE notifications
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2006-09-28 10:41:38 +00:00
Joshua Colp
dad32d9d71
I *think* this is the last list in chan_iax2
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2006-09-27 19:39:39 +00:00
Joshua Colp
a5770f4e72
Merged revisions 43783 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43792 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-09-27 17:02:28 +00:00
Matthew Fredrickson
6b3a6a1104
Fix some little things
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-09-27 17:00:48 +00:00
Matthew Fredrickson
e5435429de
Add CLI block and unblock circuit commands for SS7.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43777 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-09-27 16:49:01 +00:00
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43776 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-09-27 16:25:45 +00:00
Paul Cadach
e1e63b2355
Use ast_strdupa() instead of strdup(), thanks to sergee
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43763 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-09-27 12:32:06 +00:00
Joshua Colp
481265e347
Clean up code and convert last two things (firmware/dialplan cache) to linked list macros.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43733 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-09-27 01:39:39 +00:00
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43709 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-09-26 20:51:36 +00:00