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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44078 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44068 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If this works properly, we might have to check 1.2 to implement a backport.
The theory is that if you get a final reply in a session, it is ok to destroy the session.
If you send a final reply, you need to keep the session open for potential retransmits
from the other side. If you send a HANGUP/CANCEL, wait to the other side confirms
or until you have a timeout. If you send HANGUP/CANCEL/ACK reliably, don't destroy
the session so that you cancel the needed retransmits.
I will have to change the timer to 64*T1, but that will be a separate patch. That will
mean that if we know the roundtrip time, we can destroy sessions quicker.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@42537 65c4cc65-6c06-0410-ace0-fbb531ad65f3
now reports AST_MODULE_LOAD_DECLINE when loading if config file
is not there, also fixed an error in res_config_pgsql where it
had a non static function when it should.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41633 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- restructured build tree and makefiles to eliminate recursion problems
- support for embedded modules
- support for static builds
- simpler cross-compilation support
- simpler module/loader interface (no exported symbols)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@40722 65c4cc65-6c06-0410-ace0-fbb531ad65f3
channel to remove the theoretical race condition that the channel could get
bridged before the channel's jitterbuffer gets configured. This was pointed
out by PCadach on IRC. Thanks!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39964 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- instead of defining a free() wrapper in a bunch of files, define it as
ast_free() in utils.h and remove the copies from all the files.
- centralize and abstract the code used for doing thread storage. The code
lives in threadstorage.h, with one function being implemented in utils.c.
This new API includes generic thread storage as well as special functions
for handling thread local dynamic length string buffers.
- update ast_inet_ntoa() to use the new threadstorage API
- update ast_state2str() to use the new threadstorage API
- update ast_cli() to use the new threadstorage API
- Modify manager_event() to use thread storage. Instead of using a buffer of
4096 characters as the workspace for building the manager event, use a thread
local dynamic string. Now there is no length limitation on the length of the
body of a manager event.
- Significantly simplify the handling of ast_verbose() ...
- Instead of using a static char buffer and a lock to make sure only one
thread can be using ast_verbose() at a time, use a thread local dynamic
string as the workspace for preparing the verbose message. Instead of
locking around the entire function, the only locking done now is when the
message has been built and is being deliviered to the list of registered
verbose message handlers.
- This function was doing a strdup() on every message passed to it and
keeping a queue of the last 200 messages in memory. This has been
completely removed. The only place this was used was that if there were
any messages in the verbose queue when a verbose handler was registered,
all of the messages in the queue would be fed to it. So, I just made sure
that the console verbose handler and the network verbose handler (for
remote asterisk consoles) were registered before any verbose messages.
pbx_gtkconsole and pbx_kdeconsole will now lose a few verbose messages at
startup, but I didn't feel the performance hit of this message queue was
worth saving the initial verbose output for these very rarely used modules.
- I have removed the last three arguments to the verbose handlers, leaving
only the string itself because they aren't needed anymore. For example,
ast_verbose had some logic for telling the verbose handler to add
a newline if the buffer was completely full. Now that the buffer can grow
as needed, this doesn't matter anymore.
- remove unused function, ast_verbose_dmesg() which was to dispatch the
message queue
- Convert the list of verbose handlers to use the linked list macros.
- add missing newline characters to a few ast_verbose() calls
- convert the list of log channels to use the linked list macros in logger.c
- fix close_logger() to close all of the files it opened for logging
- update ast_log() to use a thread local dynamic string for its workspace
for preparing log messages instead of a buffer of size BUFSIZ (8kB on my
system) allocated on the stack. The dynamic string in this case is limited
to only growing to a maximum size of BUFSIZ.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r38903 | russell | 2006-08-05 01:07:39 -0400 (Sat, 05 Aug 2006) | 2 lines
suppress a compiler warning about the usage of a potentially uninitialized variable
........
r38904 | russell | 2006-08-05 01:08:50 -0400 (Sat, 05 Aug 2006) | 10 lines
Fix an issue that would cause a NewCallerID manager event to be generated
before the channel's NewChannel event. This was due to a somewhat recent
change that included using ast_set_callerid() where it wasn't before. This
function should not be used in the channel driver "new" functions.
(issue #7654, fixed by me)
Also, fix a couple minor bugs in usecount handling. chan_iax2 could have
increased the usecount but then returned an error. The place where chan_sip
increased the usecount did not call ast_update_usecount()
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r38611 | kpfleming | 2006-07-31 16:14:11 -0500 (Mon, 31 Jul 2006) | 4 lines
don't reissue hangup requests for SIP channels that have expired their RTP timeouts (one time is enough)
don't rescan the SIP private structure list too fast, it can cause channels to not be able to hang up (issue #7495, and probably others)
use ast_softhangup_nolock() since we already hold the channel's lock
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38612 65c4cc65-6c06-0410-ace0-fbb531ad65f3
int to string or string to int operations.
"pure" essentially says that this function has no side effects aside from its
result, and the result depends on nothing else other than its arguments and
global variables. "const" is a more strict form of "pure", where the function
also doesn't access any global variables.
From the gcc manual: "Such a function can be subject to common subexpression
elimination and loop optimization just as an arithmetic operator would be."
This also tells the compiler that it is safe to call the function fewer times
than the code says to, given the same arguments, since the result will always
be the same.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38452 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r38420 | file | 2006-07-28 14:49:00 -0400 (Fri, 28 Jul 2006) | 2 lines
Make a copy of the request URI in check_user_full instead of modifying the one on the structure, and also strip params properly from the user portion of the SIP URI so as to preserve the domain (issue #7552 reported by dan42)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38421 65c4cc65-6c06-0410-ace0-fbb531ad65f3
and fix a couple little things in passing
- usecnt was not initialized in chan_iax2
- ast_update_use_count() was not called after incrementing the count in chan_sip
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38077 65c4cc65-6c06-0410-ace0-fbb531ad65f3
inet_ntoa, which uses thread specific data (aka thread local storage) instead
of stack allocatted buffers to store the result.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
r36725 | russell | 2006-07-03 00:19:09 -0400 (Mon, 03 Jul 2006) | 4 lines
use ast_set_callerid to be more consistent and to make sure that the
"callerid" option in the conf files is always handled the same way and sets ANI
(issue #7285, gkloepfer)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36726 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- we need to investigate support for refusing offers of 30 ms (like the Nokia
e-series) or supporting it.
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in RFCs, as they keep referring to each other in a circular pattern in regards
to this item, but since the Nokia SIP/GSM phones use this, we might as well
start supporting it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36252 65c4cc65-6c06-0410-ace0-fbb531ad65f3
allocated. These changes caused crashes when using a channel type that did
not support the jitterbuffer. Instead of fixing why it's crashing, I'm going
to implement this in a better way next week. The way I did it caused a
jitterbuffer to be allocated on every channel where the channel type supported
jitterbuffers, even if they were disabled.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@35746 65c4cc65-6c06-0410-ace0-fbb531ad65f3
so that channels not using a jitterbuffer don't waste as much memory
- ensure that the channel drivers that use jitterbuffers can handle a failure
from configuring a jitterbuffer on a new channel because of a memory
allocation error
- On passing through these channel drivers, configure the jitterbuffer before
starting the PBX thread instead of afterwards. If the pbx fails to start for
whatever reason, this would have caused a crash.
- Also on passing, move the increase of the usecount to after all of the
possible failure conditions in the function
- fix a place where ast_update_use_count() was not called
- ensure that the owner channel pointer of the channel pvt strcutures is set to
NULL in failure conditions
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@35553 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Block fix from 1.2
- Implement part of that fix that was not already implemented, but in a different way
basically, don't cancel destruction when we receive re-transmits.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@35059 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Bug reported in the t38 issue report, but by mistake ignored before commit.
Thanks to everyone informing me about this, and Corydon for helping me sort
out sscanf :-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@34217 65c4cc65-6c06-0410-ace0-fbb531ad65f3
bug 5090 by josh colp. Thanks to everyone who help get this patch through
especially to the author Steven Underwood.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@33890 65c4cc65-6c06-0410-ace0-fbb531ad65f3
created as global variables. (The fact that these were getting created on
my system probably means that these are in the wrong place so oej, you may
want to look at this again.)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@33706 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Change variable name to make register_verify more readable (p -> peer not pvt in this function)
- Get Contact: header only once instead of twice
- Add some comments to register_verify
Caused by issue #7327... :-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@33614 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Change/add comments
- Declare internal function as static
- Remove functionname: in descriptions of functions
- Move Enums to top of file
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and some patches (all disclaimed).
- Don't change RTP properties if we reject a re-INVITE
- Don't add video to an outbound channel if there's no video on the inbound channel
- Don't include video in the "preferred codec" list for codec selection
- Clean up and document code that parses and adds SDP attachments
Since we do not transcode video, we can't handle video the same way as audio. This is a
bug fix patch. In future releases, we need to work on a solution for video negotiation,
not codecs but formats and framerates instead.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32597 65c4cc65-6c06-0410-ace0-fbb531ad65f3
fix indentation on one line;
mark XXX some unreachable code;
mark XXX another place where we could reduce the nesting depth.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31872 65c4cc65-6c06-0410-ace0-fbb531ad65f3
replacement (leave the original in case my code does not
do what the function was meant to do).
oej, please check this...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31843 65c4cc65-6c06-0410-ace0-fbb531ad65f3
and mgcp channels
- remove the jitterbuffer configuration from the pvt structures in
the sip, zap, and skinny channel drivers, as copying the same global
configuration into each pvt structure has no benefit.
- update and fix some typos in jitterbuffer related documentation
(issue #7257, north, with additional updates and modifications)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- improved support of attended transfers (REFER with replaces)
- support of INVITE/replaces in the context of a transfer
- improved support of blind transfers (REFER)
Thanks to Voop, Nuvio and Foniris for sponsoring this work.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31274 65c4cc65-6c06-0410-ace0-fbb531ad65f3
a new implementation of a fixed size jitterbuffer, as well as support for the
existing adaptive jitterbuffer implementation. (issue #3854, Slav Klenov)
Thank you very much to Slav Klenov of Securax and all of the people involved
in the testing of this feature for all of your hard work!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
use an enum for authentication results and clean up code
fix a bug where SUBSCRIBE for an unknown user/peer would not generate a response
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29903 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Joshua Colp
Kevin P. Fleming
Mark Spencer
Luigi Rizzo
Paul Cadach
Russell Bryant
Tilghman Lesher
Matt O'Gorman
Olle Johansson
BJ Weschke
North Antara