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1038 Commits

Author SHA1 Message Date
Russell Bryant
71275050ab Increase the maximum number of manager headers to 128, at the request of Pari. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55590 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-20 19:57:07 +00:00
Russell Bryant
137835c878 If the pg_config application is found, but there is probably executing it, 
then consider postgres unavailable.  (issue #8637)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-17 00:40:34 +00:00
Russell Bryant
3ed86f887e Fix the documentation on the return values from device state provider 
registration and deletion.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54218 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-13 20:56:50 +00:00
Russell Bryant
913948066e Change ast_set_state_callback() to ast_dial_set_state_callback() 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54103 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-12 19:17:08 +00:00
Russell Bryant
5bc6ee1714 - Add the ability to register a callback to monitor state changes in an 
asynchronous dial operation.
- Rename the various references to "status" to "state" in the dial API


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54066 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-12 17:58:43 +00:00
Russell Bryant
7ee02f585d Merge team/russell/sla_rewrite 
This is a completely new implementation of the SLA functionality introduced in
Asterisk 1.4.  It is now functional and ready for testing.  However, I will be
adding some additional features over the next week, as well.

For information on how to set this up, see configs/sla.conf.sample 
and doc/sla.txt.

In addition to the changes in app_meetme.c for the SLA implementation itself,
this merge brings in various other changes:

chan_sip:
 - Add the ability to indicate HOLD state in NOTIFY messages.
 - Queue HOLD and UNHOLD control frames even if the channel is not bridged to
   another channel.

linkedlists.h:
 - Add support for rwlock based linked lists.

dial.c:
 - Add the ability to run ast_dial_start() without a reference channel to
   inherit information from.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53810 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-10 00:35:09 +00:00
Russell Bryant
ff1ca74145 When we are checking for a system installed version of libgsm, we need to check 
for gsm.h as well.  Furthermore, when checking for this header, it may be
located in a gsm/ sub directory, so check for that, as well.
(issue #8773)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52997 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-30 23:23:24 +00:00
Russell Bryant
824bed6260 Clean up a few things in the last commit to the adaptive jitterbuffer code. 
- Specifically indicate to the compiler that the "dropem" variable only
   needs one but.
 - Change formatting to conform to coding guidelines.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52506 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-29 16:54:27 +00:00
Jim Dixon
2132e4f865 Fixed problem with jitterbuf, whereas it would not complain about, and 
would allow itself to be overfilled (per the max_jitterbuf parameter). Now
it rejects any data over and above that size, and complains about it.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52494 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-29 04:18:36 +00:00
Russell Bryant
6abcb7ae23 Fix the formatting of doxygen comments to properly indicate that the comment 
documents the previous entity, as opposed to the next one.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52107 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-24 21:42:47 +00:00
Joshua Colp
8acccb9254 Merge in dialing API and the app_page that uses it. (issue #BE-118) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52049 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-24 18:20:05 +00:00
Russell Bryant
33235b40d6 Merge the changes from the /team/group/vldtmf_fixup branch. 
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged.  So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio.  However,
since there was no audio coming in, the DTMF_END was never generated.  This
caused DTMF based features to no longer work.

To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf).  If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.

Channel drivers also now get passed the length of the digit to their digit_end
callback.  This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.

(issue #8597, maybe others...)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-19 17:49:38 +00:00
Kevin P. Fleming
dd357a71a7 use the ACX_PTHREAD macro from the Autoconf macro archive for setting up compiler pthreads support... should improve portability to platforms with unusual pthreads requirements 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@50867 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-15 15:03:06 +00:00
Joshua Colp
240ca25bea Add support to see whether NAT was detected (yay symmetric RTP) and also add a check in chan_sip so that if NAT has been detected and the reinvite behind nat option has been turned off, then just do partial bridge. (issue #8655 reported by mnicholson) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@50466 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-11 05:19:39 +00:00
Kevin P. Fleming
444adcb477 reduce stack consumption for AMI and AMI/HTTP requests by nearly 20K in most cases 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49676 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-05 22:16:33 +00:00
Kevin P. Fleming
46d91e71c5 add support for tracking thread-local-storage objects that exist via 'threadstorage' CLI commands 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49553 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-04 22:51:01 +00:00
Joshua Colp
345968e6fb Backport support for read/write locks. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49022 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-28 19:43:15 +00:00
Steve Murphy
4d6a91eef0 removed <err.h> as in trunk from the ael stuff. Also, threw in a minor fix to frame.c to avoid build-killing compiler warnings. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49020 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-28 19:21:56 +00:00
Kevin P. Fleming
3307ae060a move extern declaration for this option to a header file where it belongs 
provide an initial value for 'languageprefix' option, instead of relying on randomness to provide a useful value


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48998 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-27 21:08:30 +00:00
Kevin P. Fleming
b2c8abbc6d allow 'show memory' and 'show memory summary' to distinguish memory allocations that were done for caching purposes, so they don't look like memory leaks 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48987 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-27 18:29:13 +00:00
Joshua Colp
9cc04e026d Add an API call that initializes an RTP structure. We need this because chan_sip is cheeky and uses a temporary RTP structure for codec purposes, and the API calls that are used rely on the lock. (Pointed out on asterisk-dev by Andy Wang) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48964 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-26 04:31:58 +00:00
Luigi Rizzo
f9e3c1ecb0 unbreak the macro used for incrementing the frame counters. 
I don't know when the bug was introduced, but with the typical usage

	c->fin = FRAMECOUNT_INC(c->fin)

the frame counters stay to 0.

affects trunk as well (fix coming).



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48566 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-18 17:23:29 +00:00
Kevin P. Fleming
ee8ce744c3 use m4 quoting for AC_MSG_NOTICE calls, to keep these calls from thinking they have multiple arguments 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-16 21:34:41 +00:00
Kevin P. Fleming
be1b5dab06 since we really, really have to have autoconfig.h included before all other headers (especially system headers), the Makefile will now force it to happen (this will fix build problems with files like ast_expr2f.c, where we can't control the inclusion order in the file itself) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48521 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-16 20:12:41 +00:00
Joshua Colp
0995fb8aeb Payload values on the RTP structure can change AFTER a bridge has started. This comes from the packet handling of the SIP response when indication that it was answered has been sent. Therefore we need to protect this data with a lock when we read/write. (issue #8232 reported by tgrman) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48472 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-14 17:36:12 +00:00
Olle Johansson
f89143bd13 - Disable RTP hold timers while T.38 fax transmission happens 
- Encapsulate RTP timers in the rtp structure so we have one for video and one for audio
   The video one is not used in 1.4, really. Will be used for RTP keepalives when we can send
   something that video phones support in the RTP stream.
   I now this is a big architectual change at this stage for 1.4, but decided it was needed
   to avoid future bug reports.
- Document the RTP NAT keepalive option in sip.conf.sample

Issue 7679 in the bug tracker. Please test.



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48199 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-02 11:32:51 +00:00
Russell Bryant
1298cf0ea6 Backport the comment containing the warning regarding the limitations on the 
usage of this function.  It is thread safe, but not technically reentrant.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48195 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-02 03:50:58 +00:00
Joshua Colp
b2b70adede Do not do a partial bridge for Google Talk since we need to handle STUN. (issue #8448 reported by phsultan) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48168 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-11-30 21:18:24 +00:00
Joshua Colp
335630b10c Use a separate variable in the channel structure to store the context that the channel was dialed from. (issue #8382 reported by jiddings) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47850 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-11-20 15:51:37 +00:00
Matt O'Gorman
5b02ba2bf1 woohoo safe out put! 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47490 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-11-11 02:04:28 +00:00
Steve Murphy
517978fd5f These mods are to solve the problem in bug 7506. It's a lot of rework to solve a fairly small problem... such is life. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47303 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-11-07 23:46:41 +00:00
Kevin P. Fleming
f532d2f198 add an API so that translators can activate/deactivate themselves when needed 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46714 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-10-31 21:47:48 +00:00
Kevin P. Fleming
160a0448c2 revert changes that were the wrong way to address this... proper fix coming 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46713 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-10-31 21:36:17 +00:00
Kevin P. Fleming
59186bb2d2 don't re-do setup operations for translators that can dynamically register themselves 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46711 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-10-31 21:23:06 +00:00
Olle Johansson
d318976f4d Issue #8089 - Fix the ENUM support (picking one record by number). Thanks otmar! 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46631 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-10-31 10:56:20 +00:00
Kevin P. Fleming
c874ff482b ensure that items removed from a list are always unlinked from the list (next pointer set to NULL) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46511 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-10-30 21:46:07 +00:00
Olle Johansson
86c973f71f Issue #8246 - Doxygen fixes from kshumard. 
An extra big thankyou is given to everyone that contributes to doxygen!

		THANK YOU!



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46433 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-10-30 16:27:34 +00:00
Kevin P. Fleming
6c17f1e07e add passthrough and file format support for G.722 16KHz audio (issue #5084, original patch by andrew, updated by mithraen) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46154 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-10-25 00:26:17 +00:00
Kevin P. Fleming
d2b10d5f4f add an API call to allow channel drivers to determine which media formats are compatible (passthrough or transcode) with the format an existing channel is already using 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-10-24 03:45:42 +00:00
Kevin P. Fleming
227d415709 optimize the 'quick response' code a bit more... no more malloc() or memset() for each response 
expand stringfields API a bit to allow reusing the stringfield pool on a structure when needed, and remove some unnecessary code when the structure was being freed


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@45408 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-10-17 22:24:10 +00:00
Kevin P. Fleming
1944a9a07c use a configure script test for PMTU discovery control instead of just assuming it's available on Linux 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@45027 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-10-13 15:41:14 +00:00
Kevin P. Fleming
09f6a6a167 Merged revisions 44955 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r44955 | kpfleming | 2006-10-12 13:31:26 -0500 (Thu, 12 Oct 2006) | 2 lines

ensure that IAX2 and SIP sockets allow UDP fragmentation when running on Linux (thanks to Brian Candler on the asterisk-dev list for the tip)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44956 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-10-12 18:38:51 +00:00
Paul Cadach
53024e3508 CHANNEL() function sometime mix parameter and value 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44809 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-10-10 16:44:54 +00:00
Kevin P. Fleming
f804e2f153 ensure that mutex locks inside list heads are initialized properly on platforms that require constructor initialization (issue #8029, patch from timrobbins) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44631 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-10-06 21:28:03 +00:00
Kevin P. Fleming
5c4434d0e7 make LOW_MEMORY builds actually work 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44390 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-10-04 21:04:21 +00:00
Kevin P. Fleming
ff05bf15c8 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


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-10-04 19:47:22 +00:00
Kevin P. Fleming
fc72291807 ensure that local include files are always used 
avoid a duplicate function name (term_init())


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44322 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-10-04 00:25:44 +00:00
Matt O'Gorman
5058b9e13f updated res_jabber for even better component support, soon will be jep-0100 compliant. 
also removed chan_jingle and infromed info from jingle.txt, chan_gtalk still works and should be used in this version.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44240 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-10-03 00:01:34 +00:00
Kevin P. Fleming
69528cec15 eliminate compiler warning when DEBUG_CHANNEL_LOCKS is enabled and users of this header file don't also include channel.h 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43952 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-09-28 22:42:18 +00:00
Joshua Colp
d2d4833b79 Put in missing \ns on the end of ast_logs (issue #7936 reported by wojtekka) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@43933 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-09-28 18:05:43 +00:00