Commit graph

3546 commits

Author SHA1 Message Date
Luigi Rizzo
06fab1382b clarify the use of the standard SIP port number, 5060,
and rename the old DEFAULT_SIP_PORT as STANDARD_SIP_PORT
to make it clear that this is not something we can change,
unlike other defaults.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44774 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-10 08:08:35 +00:00
Luigi Rizzo
15fcc45c4e improve formatting of SIP packets when dumped to the verbose
output stream, so it is easier to find them in the log.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44770 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-10 07:40:27 +00:00
Jason Parker
a35c0989dd Merged revisions 44764 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r44764 | qwell | 2006-10-09 11:12:35 -0500 (Mon, 09 Oct 2006) | 4 lines

Fix a problem where phones that go "missing" never got unregistered.

Issue #8067, reported by pj, patch by Anthony LaMantia (with minor whitespace modifications)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44765 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-09 16:15:16 +00:00
Joshua Colp
7b3665596f Merged revisions 44759 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r44759 | file | 2006-10-09 11:41:28 -0400 (Mon, 09 Oct 2006) | 2 lines

Properly avoid a collision with iax2_hangup (issue #8115 reported by vazir)

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2006-10-09 15:49:45 +00:00
Olle Johansson
d766095219 Being pedantic... "media" is easier to understand than "data" in the function name... :-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44753 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-09 11:20:02 +00:00
Luigi Rizzo
853f7ed660 slightly restructure sipsock_read() removing a "goto"
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44752 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-09 09:04:30 +00:00
Luigi Rizzo
69af2cea2f use S_OR in one place
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44751 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-09 08:33:52 +00:00
Luigi Rizzo
ad63b4c7c1 update_call_counter(): indentation fixes and small simplifications
at the top of the function.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44750 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-08 23:26:10 +00:00
Luigi Rizzo
06ef2c1907 localize some variables and reduce nesting depth
(indentation will be fixed by a separate commit).



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44749 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-08 23:04:12 +00:00
Luigi Rizzo
dd2121b075 small simplification to initreqprep()
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44748 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-08 22:45:01 +00:00
Luigi Rizzo
36e4f62e7c Simplify function parse_request() using a single loop instead
of two very similar ones.



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2006-10-08 22:20:35 +00:00
Luigi Rizzo
b90f91c3b4 do not dereference p if we know it is NULL.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44745 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-08 14:10:26 +00:00
Olle Johansson
1b64c41369 Fix some debug output for setsockopt for TOS
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-07 20:42:27 +00:00
Olle Johansson
259f0d76b6 - move definition of global_autoframing to the same place as other globals
- set initial value at load/reload
- Add questionmarks for someone to fill in for doxygen 


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2006-10-07 18:57:11 +00:00
Olle Johansson
12ef70f0c0 Add/change doxygen and comments
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44708 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-07 16:32:31 +00:00
Olle Johansson
9cc9c38250 Explain usage of DEFAULT_SIP_PORT
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44706 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-07 16:03:36 +00:00
Olle Johansson
64aae4a887 Do *NOT* use DEFAULT_SIP_PORT in these comparisions, since users may change
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
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44674 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-07 12:20:16 +00:00
Luigi Rizzo
bc1e5f77af put common code in a function to avoid repetitions.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44664 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-07 11:28:38 +00:00
Luigi Rizzo
b37dc86c74 remove hardwired usage of 5060, use DEFAULT_SIP_PORT instead
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44663 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-07 11:11:39 +00:00
Luigi Rizzo
e691246b6e improve and document function get_in_brackets(), introducing
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
2006-10-07 09:36:09 +00:00
Luigi Rizzo
6466c2ebf9 when possible, use ast_set2_flags instead of ast_set/ast_clr .
Also mark XXX some dubious places.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44641 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-07 08:56:59 +00:00
Luigi Rizzo
ccca5843fd Two things:
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44545 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-06 09:01:09 +00:00
Joshua Colp
0b26b56316 Merged revisions 44486 via svnmerge from
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
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
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
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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2006-10-04 21:20:55 +00:00
Kevin P. Fleming
3c876af5cf Merged revisions 44378 via svnmerge from
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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2006-10-04 19:51:38 +00:00
Matt O'Gorman
7294ba3852 Merged revisions 44312 via svnmerge from
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
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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2006-10-03 18:47:38 +00:00
Joshua Colp
6727e8f599 Merged revisions 44284 via svnmerge from
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.
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
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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2006-10-02 20:13:26 +00:00
Paul Cadach
b018f59307 Merged revisions 44166 via svnmerge from
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
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
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
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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2006-09-30 04:41:04 +00:00
Kevin P. Fleming
e92c34cc38 Merged revisions 44055 via svnmerge from
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
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
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
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
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
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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
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
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-09-28 17:32:46 +00:00