some fixes for PMP Hold/Retrieve, it should work now, when briding=no
........
r61770 | crichter | 2007-04-24 15:50:05 +0200 (Di, 24 Apr 2007) | 1 line
added lock for sending messages to avoid double sending. shuffled some empty_chans after the cb_event calls, this avoids that a release_complete from a quite different call releases a fresh created setup by accident.
........
r62885 | crichter | 2007-05-03 15:59:00 +0200 (Do, 03 Mai 2007) | 1 line
fixed the problem that misdn_write did not return -1 when called with 0 samples in a frame this resultet in a deadlock in some circumstances, when the call ended because of a busy extension. added encoding of keypad.
........
increase reliability and efficiency of static Realtime config loading via ODBC:
don't request fields we aren't going to use
don't request sorting on fields that are pointless to sort on
explicitly request the fields we want, because we can't expect the database to always return them in the order they were created
(reported by blitzrage in person (!), patch by me)
improve static Realtime config loading from PostgreSQL:
don't request sorting on fields that are pointless to sort on
use ast_build_string() instead of snprintf()
don't request the list of fieldnames that resulted from the query when we both knew what they were before we ran the query _AND_ we aren't going to do anything with them anyway
(patch by me, inspired by blitzrage's bug report about res_config_odbc)
Russell Bryant [Wed, 2 May 2007 22:59:09 +0000 (22:59 +0000)]
Merge changes from team/russell/inband_dtmf ...
Fix some issues related to generating inband DTMF. There are two changes here:
1) The list of DTMF tones in the senddigit_begin() function explicitly
specified 100ms of the tone followed by 100ms of silence. This really
broke things with the way that Asterisk now wants complete control
over when the digit begins and ends. So, regardless of what Asterisk
really wanted to do, this was going to play out the tone at the length it
wanted to. This caused various problems like DTMF translation to inband to
be extremely unreliable.
The list of tones has been changed so that the correct DTMF tone is played
indefinitely until Asterisk tells it to stop.
2) ast_write() had to be modified to let a DTMF_END frame get processed even
when a generator is present. This is how the tone will finally get stopped.
(issues #8944, #9250, #9348, maybe others. Thanks to mdu113 from #8944 for
the testing and feedback!)
Issue 9638 - if a text frame is sent with no terminating NULL through a bridged
IAX connection, the remote end will receive garbage characters tacked onto the
end.
Steve Murphy [Wed, 2 May 2007 17:10:50 +0000 (17:10 +0000)]
a)In chan_zap, set the clid, src fields in channel_alloc call. b)in the channel_alloc func, set the cid_num and name fields from the arglist[blush]. c) don't update the channel app & app data fields if you are in the 'h' extension. d)the load_module func in cdr_radius needs to return DECLINE, SUCCESS.
Russell Bryant [Tue, 1 May 2007 21:57:10 +0000 (21:57 +0000)]
Merged revisions 62547 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r62547 | russell | 2007-05-01 16:55:19 -0500 (Tue, 01 May 2007) | 4 lines
Remove an unnecessary check that makes it so if you hang up after doing an
attended transfer before the target extension answers the channel, the transfer
is not successful. (issue #9338, patch by svanlund)
This patch fixes an issue where depending on the cause code, when the network
sends a PRI disconnect, the call may not be properly hung up.
(issue #9588, reported and patched by softins)
Russell Bryant [Mon, 30 Apr 2007 15:25:31 +0000 (15:25 +0000)]
When serving dynamic content, include a Cache-Control header to instruct the
browsers to not store the resulting content.
(issue #9621, reported by Pari, patch by me)
Russell Bryant [Fri, 27 Apr 2007 21:10:51 +0000 (21:10 +0000)]
Fix a weird problem where when a caller talking to someone sitting behind an
agent channel sent a digit, the digit would be played to the agent for forever.
This is because chan_agent always returned -1 from its send_digit_begin and _end
callbacks. This non-zero return value indicates to the Asterisk core that it
would like an inband DTMF generator put on the channel. However, this is the
wrong thing to do. It should *always* return 0, instead. When the digit begin
and end functions are called on the proxied channel, the underlying channel
will indicate whether inband DTMF is needed or not, and the generator will be
put on that one, and not the Agent channel.
(issue #9615, #9616, reported by jiddings and BigJimmy, and fixed by me)
Russell Bryant [Fri, 27 Apr 2007 16:14:11 +0000 (16:14 +0000)]
If no variables were passed into pbx_substitute_variables_helper_full(), then
don't even bother creating a temporary bogus channel, since that is only for
allowing certain functions to operate on the variables as if they were on a
channel. Most importantly, this fixes a crash.
(issue #9613, reported by callguy, fixed by me)
Issue #7351 - SIP Cancel fails due to the wrong contact uri. Reported by PPYY, failed to fix by OEJ
final fix by wojtekka - THANKS!!!! THis was a hard one to catch.
Revert previous fix for when the IAX2 channel goes funky (that's the technical term). This is causing legit calls to be prematurely hung up. (issue #9600 reported by justdave)
Merge rewritten group counting support. No more storing data on the variable list of the channels. That was bad, mmmk? (issue #7497 reported by sabbathbh)
Don't crash if a manager connection provides a username that exists in
manager.conf but does not have a password, and also requests MD5
authentication. (ASA-2007-012)
Russell Bryant [Tue, 24 Apr 2007 19:00:06 +0000 (19:00 +0000)]
Improve DTMF handling in ast_read() even more in response to a discussion on
the asterisk-dev mailing list. I changed the enforced minimum length of a
digit from 100ms to 80ms. Furthermore, I made it now enforce a gap of 45ms in
between digits. These values are not configurable in a configuration file
right now, but they can be easily changed near the top of main/channel.c.
Russell Bryant [Tue, 24 Apr 2007 16:16:41 +0000 (16:16 +0000)]
Add a few more state changes in handle_frame_ownerless() so that the SLA code
will get notified of these changes even when an owner channel is not provided.
This isn't from a specific bug report, it's just something I noticed while
poking around.
Russell Bryant [Mon, 23 Apr 2007 18:17:00 +0000 (18:17 +0000)]
Some dialplan functions, such as CUT(), expect to operate on variables on a
channel. So, this little hack lets them work in places where a channel doesn't
exist, such as within DUNDi configuration.
(issue #9465, reported and patched by Corydon76, testing by blitzrage)
Russell Bryant [Mon, 23 Apr 2007 17:57:32 +0000 (17:57 +0000)]
Ensure that digits passing through Asterisk have a reasonable minimum length.
It is currently 100 ms. If someone thinks this should be different, feel free
to speak up. (related to issues #8944, #9250, and #9348)
If the '* to hangup' option is not enabled, we don't need to disable * as a valid exit key.
If it was enabled, this statement would've never been checked in the first place.
Russell Bryant [Fri, 20 Apr 2007 18:19:18 +0000 (18:19 +0000)]
Fix the UpdateConfig manager action to properly treat "variables" and "objects"
differently (a=b versus a=>b).
(issue #9568, reported by pari, patch by me)
Bug 9557 - Specifying the GetVar AMI action without a Channel parameter can
cause Asterisk to crash. The reason this needs to be fixed in the functions
instead of in AMI is because Channel can legitimately be NULL, such as when
retrieving global variables.
Add OUTBOUND_GROUP_ONCE variable to app_dial. This behaves the same as OUTBOUND_GROUP except it will get unset after use so it won't get accidentally inherited. (issue #BE-140)
For those very verbose SIP implementations that attach tons of info to the Contact header... let's increase our variable sizes. (issue #9535 reported by jeffg)
If someone sets the "useragent" option in sip.conf to be empty, then don't add
the User-Agent header at all. It is an optional header, anyway. Also, the bug
report says that some of Japan's SIP providers don't allow it for some weird
reason. (issue #9488, reported by makoto, fixed by me)
Fix a bug with switching between host=dynamic and using specific hosts for
peers. The code would only reset the peer's address when it is dynamic if
it was a new peer structure. Now, it will also reset the address if it was
already in the peer list, but before the reload, it was not dynamic.
(issue #9515, reported by caio1982, fixed by me)
Remove the attempt at reporting configuration errors in sip.conf. This can
cause a bunch of improper messages when using realtime. I give up. As oej
tried to convince me when I put this in, there is just no easy way to do it.
(inspired by a message on the -dev list)
Russell Bryant [Tue, 10 Apr 2007 16:05:55 +0000 (16:05 +0000)]
File upload support was added to solve some needs for the Asterisk GUI.
However, after much discussion, it has been decided that adding this to 1.4 is
not in the best interests of the project. It has been removed here, but will
remain in trunk.
Steve Murphy [Tue, 10 Apr 2007 05:18:26 +0000 (05:18 +0000)]
Finished up a previous fix to overcome a compiler warning; the app NoCDR() has been updated to mark the channel CDR as POST_DISABLED instead of destroying the CDR; this way its flags are propagated thru a bridge and the CDR is actually dropped. The cases where only one channel in a bridge has a CDR was cleaned up.
Don't check for error when lowering priority (according to the manpage, it should never happen anyway). It might could happen, though, if another thread messed with the priority, so safeguard against that (reported via -dev list).
When calling a device that then forwards us elsewhere... we have to make our channels compatible if it is the only channel being dialed. (issue #9445 reported by marcelbarbulescu)
Russell Bryant [Fri, 6 Apr 2007 20:58:43 +0000 (20:58 +0000)]
To be able to achieve the things that we would like to achieve with the
Asterisk GUI project, we need a fully functional HTTP interface with access
to the Asterisk manager interface. One of the things that was intended to be
a part of this system, but was never actually implemented, was the ability for
the GUI to be able to upload files to Asterisk. So, this commit adds this in
the most minimally invasive way that we could come up with.
A lot of work on minimime was done by Steve Murphy. He fixed a lot of bugs in
the parser, and updated it to be thread-safe. The ability to check
permissions of active manager sessions was added by Dwayne Hubbard. Then,
hacking this all together and do doing the modifications necessary to the HTTP
interface was done by me.
Russell Bryant [Fri, 6 Apr 2007 18:58:46 +0000 (18:58 +0000)]
Fix a few problems with SLA. (issue #9459, reported by francesco_r, fixed by me)
* The original behavior was that if one station put a call on hold, another one
picked it up, and then hung up, the code would still consider the call on
hold by the first station, so the trunk would not be hung up. However, to
better comply with what most people seem to expect it to behave, it will now
hang up the trunk.
* Fix a problem with "barge=no". This was only intended to prevent people from
joining calls that are in progress. However, it also prevented other people
from picking up a call that was on hold. This has been fixed.
* When there are no active stations on a trunk and it is on hold, the code now
indicates the HOLD and UNHOLD conditions to the trunk channel. This allows
music on hold to be played to the trunk when it is on hold.
Steve Murphy [Thu, 5 Apr 2007 22:35:11 +0000 (22:35 +0000)]
Added some clarification to the example configs for CDRs, on how to select a backend. Also, made cdr-csv the default if you 'make samples', and no other changes.
projects / thirdparty / asterisk.git / log
