Russell Bryant [Sat, 10 Feb 2007 00:35:09 +0000 (00:35 +0000)]
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.
Russell Bryant [Sat, 10 Feb 2007 00:15:50 +0000 (00:15 +0000)]
When the Echo() application receives the digit '#', echo that back as well.
Since we already sent the BEGIN frame for that digit, it makes sense to send
the END as well.
Russell Bryant [Wed, 7 Feb 2007 23:52:45 +0000 (23:52 +0000)]
When building libdb1.a, put the additional flags needed at the beginning of
ASTCFLAGS, instead of at the end. This way, we ensure that we find the local
headers first before accidentally trying to use headers that exist in
locations specified in the ASTCFLAGS passed from the main Makefile.
(issue #8637, ovi)
Russell Bryant [Wed, 7 Feb 2007 20:07:39 +0000 (20:07 +0000)]
The clean target actually needs to run "distclean" on editline. This is
because we need to make sure that its configure script gets executed again,
because the CFLAGS we want to pass to editline may have changed.
Russell Bryant [Wed, 7 Feb 2007 17:39:31 +0000 (17:39 +0000)]
When parsing the NTP timestamp in a sender report message, you are supposed to
take the low 16 bits of the integer part, and the high 16 bits of the
fractional part. However, the code here was erroneously taking the low 16 bits
of the fractional part. It then shifted the result 16 bits down, so the result
was always zero. This fix makes it grab the appropriate high 16 bits, instead.
(issue #8991, pointed out by andre_abrantes)
Joshua Colp [Sat, 3 Feb 2007 21:05:02 +0000 (21:05 +0000)]
Make SIPDtmfMode application work with recent capability changes, and also fix an RTP stack issue when the auto option was used. (issue #8972 reported by mdu113)
Russell Bryant [Sat, 3 Feb 2007 20:44:20 +0000 (20:44 +0000)]
Merged revisions 53133 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r53133 | russell | 2007-02-03 14:38:13 -0600 (Sat, 03 Feb 2007) | 4 lines
set the DIALSTATUS variable to contain "INVALIDARGS" when the dial application
exits early because of invalid arguments instead of just leaving it empty.
(issue #8975)
BJ Weschke [Fri, 2 Feb 2007 21:26:45 +0000 (21:26 +0000)]
I'm baaaaaaaaaack. :)
Post a warning to the console that things might possibly be misconfigured when queue member's states are still 'Not in Use' when we're about to bridge them with a caller from queue. Also, put some documentation quoted from oej's queues.txt efforts started in /trunk today.
This commit puts #7433 into feedback state for 1.4, and pending no further negative feedback, it will finally be closed.
Pass the glob expanded filename to process_text_line so that error messages contain the actual filename, not the original include one. (issue #8959 reported by tzafrir)
Olle Johansson [Fri, 2 Feb 2007 00:24:03 +0000 (00:24 +0000)]
Disable the direct p2p RTP call setup in SIP. You can enable it in sip.conf, but it is now
considered experimental until we solve the AST_CONTROL_ANSWER with payload and videocaps
stuff.
Olle Johansson [Thu, 1 Feb 2007 21:05:34 +0000 (21:05 +0000)]
- Clean INC_COUNT flag when we decrement call counter
- If it's still set at time of dialog destruction, make sure we decrement the device call counter properly
before we destroy the dialog
Joshua Colp [Thu, 1 Feb 2007 00:24:20 +0000 (00:24 +0000)]
When going on hold have the side that was put on hold reinvite back to Asterisk. When going off hold have the side that was taken off hold reinvited back to the other party.
Russell Bryant [Wed, 31 Jan 2007 17:34:22 +0000 (17:34 +0000)]
Instead of always creating a realtime queue member as unpaused, read the
"paused" column and use that value for the paused status of the member.
(issue #8949, jmls)
Russell Bryant [Tue, 30 Jan 2007 23:23:24 +0000 (23:23 +0000)]
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)
Russell Bryant [Tue, 30 Jan 2007 19:43:59 +0000 (19:43 +0000)]
Blocked revisions 52954 via svnmerge
........
r52954 | russell | 2007-01-30 13:41:52 -0600 (Tue, 30 Jan 2007) | 4 lines
Don't print a message indicating that we don't know what to do with a proceeding
control frame in ast_request_and_dial(). We just need to ignore it.
(reported by JerJer on #asterisk-dev)
Russell Bryant [Tue, 30 Jan 2007 19:33:12 +0000 (19:33 +0000)]
Only set the DTMF flag on the rtp structure if the DTMF mode is actually
RFC2833, not just that it is not INFO. This makes it get set for inband DTMF
as well, which is not valid.
(issue #8936)
Russell Bryant [Tue, 30 Jan 2007 17:19:39 +0000 (17:19 +0000)]
Merged revisions 52903 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r52903 | russell | 2007-01-30 11:12:04 -0600 (Tue, 30 Jan 2007) | 9 lines
The SIGHUP handler was implemented to allow admins to send SIGHUP to a running
Asterisk process to reload the configuration. However, doing the actual reload
in the signal handler itself is a very bad thing to do, because the reload
process includes calling non-reentrant functions such as malloc/calloc/etc.
If Asterisk is running in the background, then the reload will happen
immediately. However, if running in console mode, the reload doesn't work
until something is typed at the console. That sort of defeats the purpose,
but I don't see an easy way to get around it at this point.
Russell Bryant [Tue, 30 Jan 2007 15:36:36 +0000 (15:36 +0000)]
Blocked revisions 52857 via svnmerge
........
r52857 | russell | 2007-01-30 09:35:23 -0600 (Tue, 30 Jan 2007) | 5 lines
Comment out the parts in the Makefile that make codec_zap get built. It will
not yet build against zaptel 1.2, so I am disabling it to prevent further bug
reports until it gets merged.
(issue #8940)
Russell Bryant [Tue, 30 Jan 2007 00:15:50 +0000 (00:15 +0000)]
Merged revisions 52762 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r52762 | russell | 2007-01-29 18:15:06 -0600 (Mon, 29 Jan 2007) | 5 lines
Fix the extraction of the timestamp from video frames. It was using the
mapping for a mini-frame instead of a video-frame, which caused it to
get invalid data.
(issue #8795, mihai)
Now that filename is part of the structure and since it comes before postprocess... we have to add it to our postprocess line. (reported on asterisk-dev by Boris Bakchiev)
Russell Bryant [Mon, 29 Jan 2007 21:26:27 +0000 (21:26 +0000)]
Fix a problem with packet-to-packet bridging and DTMF mode translation. P2P
bridging can only be used when the DTMF modes don't match if the core is
monitoring DTMF in both directions. Then, the core will handle the translation.
Otherwise, this bridging method can not be used.
(issue #8936)
Russell Bryant [Mon, 29 Jan 2007 20:39:20 +0000 (20:39 +0000)]
The session lock can not be held while calling action callbacks. If so, then
when the WaitEvent callback gets called, then no event can happen because the
session can't be locked by another thread. Also, the session needs to be
locked in the HTTP callback when it reads out the output string. This fixes
the deadlock reported in both 8711 and 8934.
Regarding issue 8711, there still may be an issue. If there is a second action
requested before the processing of the first action is finished, there could
still be some corruption of the output string buffer used to build the result.
(issue #8711, #8934)
Steve Murphy [Mon, 29 Jan 2007 17:57:08 +0000 (17:57 +0000)]
this is for 8778 (pt_BR backport to 1.4). It was committed to trunk via 7663. But it wasn't so much an enhancement as a fix for the bad language output for portuguese in Brazil, so, after a lot of prodding from patient Brazilians, here is the same fix for 1.4
Russell Bryant [Mon, 29 Jan 2007 16:54:27 +0000 (16:54 +0000)]
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.
Jim Dixon [Mon, 29 Jan 2007 04:18:36 +0000 (04:18 +0000)]
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.
Joshua Colp [Sat, 27 Jan 2007 00:44:51 +0000 (00:44 +0000)]
Have the manager interface send back an "Already logged in" message instead of "Invalid/Unknown Command" when the client authenticates for a second time. (issue #8509 reported by pari)
projects / thirdparty / asterisk.git / log
