Commit graph

5034 commits

Author SHA1 Message Date
Joshua Colp
10cdbe28a8 Merged revisions 109386 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r109386 | file | 2008-03-18 11:58:39 -0300 (Tue, 18 Mar 2008) | 3 lines

Put a maximum limit on the number of payloads accepted, and also make sure a given payload does not exceed our maximum value.
(AST-2008-002)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109390 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-18 15:08:09 +00:00
Jason Parker
263c658a6b Do not return with a successful authentication if the From header ends up empty.
(AST-2008-003)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109389 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-18 15:07:04 +00:00
Olle Johansson
0de4eba640 Add manager peerstatus events when peer can't authenticate.
(closes issue #11959)
Reported by: mostyn
Patches: 
      peerstatus3.patch uploaded by mostyn (license 398)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-18 07:23:45 +00:00
Michiel van Baak
5729b0ec40 Update the directory of placed calls on skinny phones
when dialing a channel that does not provide progress (analog ZAP lines)

The phone does handle the double update on calls to channels that do
provide progress and wont insert duplicate items

(closes issue #12239)
Reported by: DEA
Patches:
      chan_skinny-call-log.txt uploaded by DEA (license 3)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109168 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-17 17:43:46 +00:00
Joshua Colp
7980ac1261 Remove something that is never ever used.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109113 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-17 16:47:02 +00:00
Joshua Colp
ba63fd28c2 Merged revisions 109107 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r109107 | file | 2008-03-17 13:24:29 -0300 (Mon, 17 Mar 2008) | 4 lines

200 OKs in response to a reinvite need to be sent reliably. If the remote side does not receive one the dialog will be torn down.
(closes issue #12208)
Reported by: atrash

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109108 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-17 16:26:36 +00:00
Joshua Colp
f7f58d194e Make sure that the temporary sip_request structure is empty so that copy_request doesn't think it already has an ast_str.
(closes issue #12231)
Reported by: IgorG


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109054 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-17 14:37:40 +00:00
Russell Bryant
5bd4e32a4c Merged revisions 108796 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r108796 | russell | 2008-03-14 15:09:22 -0500 (Fri, 14 Mar 2008) | 5 lines

Fix a channel name issue.  chan_oss registers the "Console" channel type,
but it created channels with an "OSS" prefix.

(closes issue #12194, reported by davidw, patched by me)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108797 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-14 20:09:37 +00:00
Mark Michelson
086d4f0e56 Merged revisions 108737 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r108737 | mmichelson | 2008-03-14 11:44:08 -0500 (Fri, 14 Mar 2008) | 33 lines

Fix a race condition in the SIP packet scheduler which could cause a crash.

chan_sip uses the scheduler API in order to schedule retransmission of reliable
packets (such as INVITES). If a retransmission of a packet is occurring, then the
packet is removed from the scheduler and retrans_pkt is called. Meanwhile, if
a response is received from the packet as previously transmitted, then when we 
ACK the response, we will remove the packet from the scheduler and free the packet.

The problem is that both the ACK function and retrans_pkt attempt to acquire the
same lock at the beginning of the function call. This means that if the ACK function
acquires the lock first, then it will free the packet which retrans_pkt is about to
read from and write to. The result is a crash.

The solution:

1. If the ACK function fails to remove the packet from the scheduler and the retransmit
   id of the packet is not -1 (meaning that we have not reached the maximum number of 
   retransmissions) then release the lock and yield so that retrans_pkt may acquire the
   lock and operate.

2. Make absolutely certain that the ACK function does not recursively lock the lock in
   question. If it does, then releasing the lock will do no good, since retrans_pkt will
   still be unable to acquire the lock.

(closes issue #12098)
Reported by: wegbert
(closes issue #12089)
Reported by: PTorres
Patches:
      12098-putnopvutv3.patch uploaded by putnopvut (license 60)
Tested by: jvandal


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108738 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-14 16:52:51 +00:00
Russell Bryant
5f58a11ff2 Merged revisions 108530 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r108530 | russell | 2008-03-13 16:06:33 -0500 (Thu, 13 Mar 2008) | 10 lines

Make a tweak that gets the LEDs on polycom phones to blink when an extension that
has been subscribed to goes on hold.  Otherwise, they just stay on like it does
when an extension is in use.

(closes issue #11263)
Reported by: russell
Patches:
      notify_hold.rev1.txt uploaded by russell (license 2)
Tested by: russell

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108531 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-13 21:06:52 +00:00
Russell Bryant
1a2b358588 Merge changes from team/jamesgolovich/chan_sip-ast_str
This set of changes removes the hard coded maximum packet size of 4kB from chan_sip.
It now starts by allocating 1kB, and growing the buffer as needed to accommodate large
packets.

(closes issue #8556, reported by mikma, patch by jamesgolovich)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108439 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-13 19:54:44 +00:00
Russell Bryant
8bbef5f996 Rename ast_tcptls_server_instance to session_instance, since this pertains to
server and client usage.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108295 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-12 22:13:18 +00:00
Mark Michelson
ff1527de3d Let's get this to compile
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108293 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-12 22:09:52 +00:00
Mark Michelson
39cc1b4f36 Merged revisions 108288 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r108288 | mmichelson | 2008-03-12 16:53:46 -0500 (Wed, 12 Mar 2008) | 14 lines

Change AST_SCHED_DEL use to ast_sched_del for autocongestion in chan_sip.

The scheduler callback will always return 0. This means that this id 
is never rescheduled, so it makes no sense to loop trying to delete
the id from the scheduler queue. If we fail to remove the item from the
queue once, it will fail every single time.

(Yes I realize that in this case, the macro would exit early because the
id is set to -1 in the callback, but it still makes no sense to use
that macro in favor of calling ast_sched_del once and being done with it)

This is the first of potentially several such fixes.


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108289 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-12 21:57:41 +00:00
Kevin P. Fleming
a3a8aa6547 add support for named sections in zapata.conf, and fix a few bugs in config file parsing
(closes issue #9503)
Reported by: tzafrir
Patches:
      fix_cleanups uploaded by tzafrir (license 46)
      zapata_sections uploaded by tzafrir (license 46)
      skipchannel_options uploaded by tzafrir (license 46)
      conf_sample uploaded by tzafrir (license 46)

patches updated by me to better conform to coding guidelines and fix some problems



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108286 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-12 21:37:40 +00:00
Kevin P. Fleming
5ebfa5638a Merged revisions 108086 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r108086 | kpfleming | 2008-03-12 14:16:07 -0500 (Wed, 12 Mar 2008) | 6 lines

if we receive an INVITE with a Content-Length that is not a valid number, or is zero, then don't process the rest of the message body looking for an SDP

closes issue #11475
Reported by: andrebarbosa


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108191 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-12 20:27:01 +00:00
Tilghman Lesher
582d3b4ba7 Deadlock fixes
(closes issue #12143)
 Reported by: kactus
 Patches: 
       20080312__bug12143__2.diff.txt uploaded by Corydon76 (license 14)
 Tested by: kactus


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107998 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-12 07:43:03 +00:00
Jason Parker
1c0bc928d1 Merged revisions 107714 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r107714 | qwell | 2008-03-11 15:49:56 -0500 (Tue, 11 Mar 2008) | 5 lines

Copy voicemail dependency logic for res_adsi to chan_gtalk and chan_jingle (for jabber).

(closes issue #12014)
Reported by: junky

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107718 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11 20:53:48 +00:00
Kevin P. Fleming
3ee2872f40 Merged revisions 107713 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r107713 | kpfleming | 2008-03-11 15:48:58 -0500 (Tue, 11 Mar 2008) | 2 lines

get chan_vpb to build properly in dev mode

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107715 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11 20:50:57 +00:00
Kevin P. Fleming
acb73c7448 fix another potential bug found by gcc 4.3
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11 15:39:37 +00:00
Kevin P. Fleming
4925e7b835 Merged revisions 107464 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r107464 | kpfleming | 2008-03-11 09:53:03 -0500 (Tue, 11 Mar 2008) | 2 lines

fix various other problems found by gcc 4.3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107466 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11 15:13:38 +00:00
Terry Wilson
a9a3f001e0 Merged revisions 107290 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r107290 | twilson | 2008-03-10 19:59:18 -0500 (Mon, 10 Mar 2008) | 2 lines

If we fail to alloc a channel, we should re-lock the pvt structure before returning.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107292 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11 01:09:46 +00:00
Jason Parker
6477f207f1 Merged revisions 107173 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r107173 | qwell | 2008-03-10 15:27:08 -0500 (Mon, 10 Mar 2008) | 5 lines

Make sure to reenable echo can after a "failed" (canceled, etc) three-way call.

(closes issue #11335)
Reported by: rebuild

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107177 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-10 20:28:33 +00:00
Joshua Colp
362b184c9c If we receive a 488 on a T38 request reinvite back to audio. As well reinvite across a bridge back to audio if one side doesn't negotiate to T38.
(closes issue #8677)
Reported by: alex-911


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107157 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-10 20:00:21 +00:00
Kevin P. Fleming
f5bee50ad1 Merged revisions 106945 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r106945 | kpfleming | 2008-03-08 09:59:42 -0600 (Sat, 08 Mar 2008) | 2 lines

don't generate D-Channel "up" and "down" messages unless the channel state is actually changing; also, generate the "up" message when an implicit "up" occurs due to reception of a normal event when we thought the channel was "down"

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106946 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-08 16:03:48 +00:00
Matthew Fredrickson
8874940104 Make sure we don't start a call when we have already done so in response to a COT message
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106892 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-07 22:36:49 +00:00
Steve Murphy
377e51c4d4 (closes issue #6002)
Reported by: rizzo
Tested by: murf

Proposal of the changes to be made, and then an announcement of how they were accomplished:

http://lists.digium.com/pipermail/asterisk-dev/2008-February/032065.html

and:

http://lists.digium.com/pipermail/asterisk-dev/2008-March/032124.html

Here is a recap, file by file, of what I have done:

pbx/pbx_config.c
pbx/pbx_ael.c

All funcs that were passed a ptr to the context list, now will ALSO be passed a hashtab ptr to the same set.
Why? because (for the time being), the dialplan is stored in both, to facilitate a quick, low-cost move to
hash-tables to speed up dialplan processing. If it was deemed necessary to pass the context LIST, well, it
is just as necessary to have the TABLE available. This is because the list/table in question might not be
the global one, but temporary ones we would use to stage the dialplan on, and then swap into the global
position when things are ready.

We now have one external function for apps to use, "ast_context_find_or_create()" instead of the pre-existing
"find" and "create", as all existing usages used both in tandem anyway.

pbx_config, and pbx_ael, will stage the reloaded dialplan into local lists and tables, and 
then call merge_contexts_and_delete, which will merge (now) existing contexts and 
priorities from other registrars into this local set by copying them. Then, merge_contexts_and_delete will
lock down the contexts, swap the lists and tables, and unlock (real quick), and then 
destroy the old dialplan.



chan_sip.c
chan_iax.c
chan_skinny.c

All the channel drivers that would add regcontexts now use the ast_context_find_or_create now.

chan_sip also includes a small fix to get rid of warnings about removing priorities that never got entered.


apps/app_meetme.c
apps/app_dial.c
apps/app_queue.c

All the apps that added a context/exten/priority were also modified to use ast_context_find_or_create instead.


include/asterisk/pbx.h

ast_context_create() is removed. Find_or_create_ is the new method.
ast_context_find_or_create()  interface gets the hashtab added.
ast_merge_contexts_and_delete() gets the local hashtab arg added.
ast_wrlock_contexts_version() is added so you can detect if someone else got a writelock between your readlocking and writelocking.
ast_hashtab_compare_contexts was made public for use in pbx_config/pbx_ael
ast_hashtab_hash_contexts was in like fashion make public.


include/asterisk/pval.h

ast_compile_ael2() interface changed to include the local hashtab table ptr.


main/features.c

For the sake of the parking context, we use ast_context_find_or_create().



main/pbx.c

I changed all the "tree" names to "table" instead. That's because the original
implementation was based on binary trees. (had a free library). Then I moved
to hashtabs. Now, the names move forward too.

refcount field added to contexts, so you can keep track of how many modules
wanted this context to exist.

Some log messages that are warnings were inflated from LOG_NOTICE to LOG_WARNING.

Added some calls to ast_verb(3,...) for debug messages

Lots of little mods to ast_context_remove_extension2, which is now excersized in ways
it was not previously; one definite bug fixed.

find_or_create was upgraded to handle both local lists/tables as well as the globals.

context_merge() was added to do the per-context merging of the old/present contexts/extens/prios into the new/proposed local list/tables

ast_merge_contexts_and_delete() was heavily modified.

ast_add_extension2() was also upgraded to handle changes. 

the context_destroy() code was re-engineered to handle the new way of doing things,
by exten/prio instead of by context.



res/ael/pval.c
res/ael/ael.tab.c
res/ael/ael.tab.h
res/ael/ael.y
res/ael/ael_lex.c
res/ael/ael.flex
utils/ael_main.c
utils/extconf.c
utils/conf2ael.c
utils/Makefile

Had to change the interface to ast_compile_ael2(), to include the hashtab ptr.
This ended up involving several external apps.  The main gotcha was I had to 
include lock.h and hashtab.h in several places.


As a side note, I tested this stuff pretty thoroughly, I replicated the problems
originally reported by Luigi, and made triply sure that reloads worked, and everything
worked thru "stop gracefully". I found a and fixed a few bugs as I was merging into
trunk, that did not appear in my tests of bug6002.

How's this for verbose commit messages?




git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106757 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-07 18:57:57 +00:00
Tilghman Lesher
8718878490 Merged revisions 106552 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r106552 | tilghman | 2008-03-07 00:36:33 -0600 (Fri, 07 Mar 2008) | 6 lines

Safely use the strncat() function.
(closes issue #11958)
 Reported by: norman
 Patches: 
       20080209__bug11958.diff.txt uploaded by Corydon76 (license 14)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106553 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-07 06:54:47 +00:00
Joshua Colp
496adc6fc0 Merged revisions 106235 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r106235 | file | 2008-03-05 18:32:10 -0400 (Wed, 05 Mar 2008) | 4 lines

Add a control frame to indicate the source of media has changed. Depending on the underlying technology it may need to change some things.
(closes issue #12148)
Reported by: jcomellas

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106239 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-05 22:43:22 +00:00
Russell Bryant
82b3a87fd7 Merged revisions 106237 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r106237 | russell | 2008-03-05 16:37:09 -0600 (Wed, 05 Mar 2008) | 3 lines

Fix a potential deadlock and a few different potential crashes.
(closes issue #12145, reported by thiagarcia, patched by me)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106238 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-05 22:40:58 +00:00
Kevin P. Fleming
c6585e5b5d Merged revisions 106038 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r106038 | kpfleming | 2008-03-05 09:32:35 -0600 (Wed, 05 Mar 2008) | 7 lines

when a PRI call must be moved to a different B channel at the request of the other endpoint, ensure that any DSP active on the original channel is moved to the new one

(closes issue #11917)
Reported by: mavetju
Tested by: mavetju


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2008-03-05 15:40:40 +00:00
Tilghman Lesher
7a3f642207 Merged revisions 106015 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r106015 | tilghman | 2008-03-05 09:17:16 -0600 (Wed, 05 Mar 2008) | 7 lines

Correctly initialize retransid in SIP, and ensure that the warning when failing to delete a schedule entry can actually hit the log.
(closes issue #12140)
 Reported by: slavon
 Patches: 
       sch2.patch uploaded by slavon (license 288)
(Patch slightly modified by me)

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2008-03-05 15:23:32 +00:00
Tilghman Lesher
7007c565fe Fix minor misuses of snprintf
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105841 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-04 23:10:45 +00:00
Russell Bryant
96e04792bd add a destroy API call for a server instance
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105804 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-04 22:28:03 +00:00
Russell Bryant
cc55483858 More public API name changes to use an appropriate ast_ prefix
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-04 22:23:21 +00:00
Russell Bryant
efb1e30a38 Rename public object server_instance to ast_tcptls_server_instance
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105773 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-04 22:15:18 +00:00
Russell Bryant
7b1e335999 Fix some bugs in the SIP tcp helper thread.
- fix a spot where a lock wouldn't get unlocked in an error condition
 - call ast_mutex_destroy() on the lock before freeing its memory

(related to issue #11972)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105734 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-04 20:36:16 +00:00
Joshua Colp
4de0d8368f Merged revisions 105674 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r105674 | file | 2008-03-04 14:05:28 -0400 (Tue, 04 Mar 2008) | 8 lines

When a new source of audio comes in (such as music on hold) make sure the marker bit gets set.
(closes issue #10355)
Reported by: wdecarne
Patches:
      10355.diff uploaded by file (license 11)
(closes issue #11491)
Reported by: kanderson

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105675 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-04 18:08:42 +00:00
Russell Bryant
0db8a98efe Fix some code that was improperly changed in revision 104866 from issue #12079.
(closes issue #12129, reported by elguero, patched by me)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105574 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-03 18:49:34 +00:00
Russell Bryant
a584b51dd9 Merged revisions 105570 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r105570 | russell | 2008-03-03 11:16:53 -0600 (Mon, 03 Mar 2008) | 3 lines

In the case of an ast_channel allocation failure, take the local_pvt out of the
pvt list before destroying it.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105571 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-03 17:17:27 +00:00
Russell Bryant
bc3f768547 Merged revisions 105568 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r105568 | russell | 2008-03-03 11:05:16 -0600 (Mon, 03 Mar 2008) | 3 lines

Fix a potential memory leak of the local_pvt struct when ast_channel allocation
fails.  Also, in passing, centralize the code necessary to destroy a local_pvt.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105569 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-03 17:06:35 +00:00
Joshua Colp
b336ac48cf Merged revisions 105557 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r105557 | file | 2008-03-03 11:15:39 -0400 (Mon, 03 Mar 2008) | 6 lines

Add a comment to describe some logic.
(closes issue #12120)
Reported by: flefoll
Patches:
      chan_sip.c.br14.patch-just-a-comment uploaded by flefoll (license 244)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105558 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-03 15:16:57 +00:00
Philippe Sultan
7293986e44 Remove unnecessary if statements before calling iks_delete (redundant check is
done inside iks_delete), thus making the code conform with coding guidelines.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105263 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-29 14:15:03 +00:00
Tilghman Lesher
557c38bc8d Fix crash when configuration does not match hardware detection.
(closes issue #12096)
 Reported by: mmickan
 Patches: 
       chan_vpb.cc.diff uploaded by mmickan (license 400)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104974 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-28 14:42:32 +00:00
Jason Parker
decec84c56 Merged revisions 104920 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r104920 | qwell | 2008-02-27 22:31:21 -0600 (Wed, 27 Feb 2008) | 2 lines

According to a video at www.cisco.com, the 7921G supports 6 line appearances.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104921 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-28 04:37:28 +00:00
Russell Bryant
60eec5bdcb reduce indentation in alloc_sub
(issue #12079)
Reported by: tzafrir
Patches:
      alloc_sub uploaded by tzafrir (license 46)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104866 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-27 23:58:49 +00:00
Joshua Colp
1281c05afa After further discussion revert my previous commit for this. Currently in order to ensure devicestate is the expected value in another module (such as app_queue) then chan_sip must be loaded before hand.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104595 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-27 17:04:16 +00:00
Joshua Colp
5a5b0be411 When queueing up a device state change when the peer is loaded from the configuration give it a state of not in use. We have to do this because the channel technology may not yet be registered so the state could not be queried and would be considered invalid.
(closes issue #12087)
Reported by: liorm


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-27 16:26:57 +00:00
Joshua Colp
dca12f4aa7 Fix T38 passthrough regression introduced by state changes.
(closes issue #12078)
Reported by: dimas
Patches:
      v1-12078.patch uploaded by dimas (license 88)
(closes issue #12074)
Reported by: Ivan


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104533 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-27 15:31:09 +00:00
Tilghman Lesher
4aff24881b Bring Voicetronix driver up to date with current drivers
(closes issue #12084)
 Reported by: mmickan
 Patches: 
       chan_vpb.cc.diff uploaded by mmickan (license 400)
       module.h.diff uploaded by mmickan (license 400)
       vpb.conf.sample uploaded by mmickan (license 400)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104502 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-27 08:20:15 +00:00