added method standard_dec for dialing out on groups, to avoid conflicts, which caused issues with some ISDN providers
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r58850 | crichter | 2007-03-13 13:58:32 +0100 (Di, 13 Mär 2007) | 1 line
fixed the crypt_keys stuff
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r59062 | crichter | 2007-03-20 10:18:06 +0100 (Di, 20 Mär 2007) | 1 line
avoid sending a disconnect when we already received one.
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r59063 | crichter | 2007-03-20 10:23:22 +0100 (Di, 20 Mär 2007) | 1 line
Russell Bryant [Thu, 15 Mar 2007 22:25:12 +0000 (22:25 +0000)]
Merge changes from svn/asterisk/team/russell/LaTeX_docs.
* Convert most of the doc directory into a single LaTeX formatted document
so that we can generate a PDF, HTML, or other formats from this
information.
* Add a CLI command to dump the application documentation into LaTeX format
which will only be include if the configure script is run with
--enable-dev-mode.
* The PDF turned out to be close to 1 MB, so it is not included. However, you
can simply run "make asterisk.pdf" to generate it yourself. We may include
it in release tarballs or have automatically generated ones on the web site,
but that has yet to be decided.
Russell Bryant [Wed, 14 Mar 2007 19:18:08 +0000 (19:18 +0000)]
Some people like to put "limitonpeer" instead of "limitonpeers" in their
configuration. While we're at it, support "limitonpeerz" and
"limitonpeerssssss". (inspired by issue #9172)
Russell Bryant [Wed, 14 Mar 2007 16:33:01 +0000 (16:33 +0000)]
By default, don't attempt to do any CallerID handling at all with SLA because
it is known to not work properly in some situations. However, add an option to
enable it for those that would like to use it anyway.
The short story behind this is that to properly handle CallerID with SLA, we
need the ability to change the CallerID on an existing call, and we are not
ready to handle that.
Russell Bryant [Tue, 13 Mar 2007 23:19:51 +0000 (23:19 +0000)]
Ensure that the blinky lights show that the trunk stopped ringing when the
trunk hangs up before a station has answered it.
(issue #9234, reported by francesco_r)
Joshua Colp [Mon, 12 Mar 2007 01:21:12 +0000 (01:21 +0000)]
Allow RFC2833 compensation to compensate for even stupider implementations by queueing up the end frame at the start, not the actual end. (issue #8963 reported by AndrewZ)
Joshua Colp [Mon, 12 Mar 2007 00:51:16 +0000 (00:51 +0000)]
Add matchexterniplocally setting which only substitutes your externip/externhost setting if it matches the localnet setting. I know of at least two people who need opposite settings, so I made it an option! (issue #8821 reported by kokoskarokoska)
Russell Bryant [Sat, 10 Mar 2007 18:11:11 +0000 (18:11 +0000)]
Fix a few more places in chan_iax2 where the ast_frame used for receiving a
frame was not properly initialized.
- Interpolating a frame when the jitterbuffer is in use
- decrypting a frame when IAX2 encryption is on
- frames in an IAX2 trunk
Russell Bryant [Fri, 9 Mar 2007 23:59:10 +0000 (23:59 +0000)]
Merge some updates to the SLA documentation. I plan to keep working on this
to explain all of the expected behavior with call handling, configuration
details for specific phones, and other things. However, I got tired of doing
it in plain text, so I switched to using LaTeX. I have included the PDF version.
I haven't been able to get a nice looking plain text version out of it yet, but
I'm not terribly concerned since this is supposed to be more of the manual,
while the plain text sample configuration file is the reference.
If we are unable to lookup the host in a c line we have to abort, otherwise the previous data is gone and we will (potentially) have no data when all is said and done.
Russell Bryant [Thu, 8 Mar 2007 22:15:15 +0000 (22:15 +0000)]
Hang up the channel that put the call on hold in the event processing thread to
avoid a race condition. Also, if the station originated the call that it is
putting on hold, don't hang up the trunk if it was the only station on the call
and it is hanging up due to hold and not a normal hangup.
Russell Bryant [Thu, 8 Mar 2007 01:01:46 +0000 (01:01 +0000)]
If we receive ZT_EVENT_REMOVED, destroy the specified channel.
(issue #7256, tzafrir)
Also, update the configure script to make sure that we don't try to build
chan_zap if the installed version of zaptel does not include ZT_EVENT_REMOVED.
Russell Bryant [Wed, 7 Mar 2007 18:19:19 +0000 (18:19 +0000)]
(This bug was reported to me by Kinsey Moore)
Merged revisions 58242 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58242 | russell | 2007-03-07 12:17:07 -0600 (Wed, 07 Mar 2007) | 7 lines
Fix a problem where the Asterisk channel name could be that of the wrong IAX2
user for a call. This is because the first step of choosing this name is to
look for an IAX2 peer that happens to have the same IP/port number that this
call is coming from and assuming that is it. However, this is not always
correct. So, I have made it change this name after authentication happens
since at that point, we have an exact match.
Russell Bryant [Wed, 7 Mar 2007 00:25:19 +0000 (00:25 +0000)]
Merged revisions 58164 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58164 | russell | 2007-03-06 18:20:13 -0600 (Tue, 06 Mar 2007) | 4 lines
If the channels acquired using the manager Redirect action are not up, then
don't attempt to do anything with them. It could lead to weird behavior,
including crashes. (issue #8977)
Steve Murphy [Tue, 6 Mar 2007 23:10:14 +0000 (23:10 +0000)]
Merged revisions 58115 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58115 | murf | 2007-03-06 15:52:52 -0700 (Tue, 06 Mar 2007) | 1 line
Fix for 9220: Eyebeam cannot renew subscriptions for presence info. Reason: re-SUBSCRIBE requests don't include Accept headers, which the rfc says are optional (to put it tersely), (it uses MAY), and luckily, the sip_pvt struct has the format info stored, so we simply leave it if the format is set, and the accept header null.
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Steve Murphy [Fri, 2 Mar 2007 05:21:36 +0000 (05:21 +0000)]
I almost had comma escapes right, but 9184 points out the problem-- the escape is removed by pbx_config, and pbx_ael should also, before sending it down into the pbx engine. Also, you have to insert it back in, if you are generating extensions.conf code from the AEL.
Russell Bryant [Thu, 1 Mar 2007 23:42:53 +0000 (23:42 +0000)]
Merge changes from svn/asterisk/team/russell/sla_updates
* Originally, I put in the documentation that only Zap interfaces would be
supported on the trunk side. However, after a discussion with Qwell, we came
up with a way to make IP trunks work as well, using some things already in
Asterisk. So, here it is, this now officially supports IP trunks.
* Update the SLA documentation to reflect how to setup IP trunks.
* Add a section in sla.txt that describes how to set up an SLA system with
voicemail.
* Simplify the way DTMF passthrough is handled in MeetMe.
* Fix a bug that exposed itself when using a Local channel on the trunk side
in SLA. The station's channel needs to be passed to the dial API when
dialing the trunk.
* Change a WARNING message to DEBUG in channel.h. This message is of no use
to users.
Don't even attempt to optimize things when a proxy channel is involved. It will just explode in weird and unexplaineable ways. (issue #9175 reported by clegall_proformatique)
Russell Bryant [Wed, 28 Feb 2007 22:07:05 +0000 (22:07 +0000)]
Merge more changes from svn/asterisk/team/russell/sla_updates
* Add support for private hold. By setting "hold=private" for a trunk, only
the station that put the call on hold will be able to retrieve it from hold.
Also, by setting "hold=private" for a station, any call that station puts
on hold can only be retrieved by that station.
Russell Bryant [Wed, 28 Feb 2007 19:56:20 +0000 (19:56 +0000)]
Merge changes from svn/asterisk/team/russell/sla_updates
* Add support for the "barge=no" option for trunks. If this option is set,
then stations will not be able to join in on a call that is on progress
on this trunk.
Russell Bryant [Wed, 28 Feb 2007 18:20:05 +0000 (18:20 +0000)]
Merge current set of changes from svn/asterisk/team/russell/sla_updates
* Add support for station ring delays. Ring delays can be set globally for a
station or for specific trunks on the station.
* Fix a few bugs in existing code.
* Restructure and Reorganize code to improve readability and maintainability.
* Improve formatting of the "sla show (trunks|stations)" CLI commands.
Steve Murphy [Wed, 28 Feb 2007 17:15:27 +0000 (17:15 +0000)]
I was surprised that I had not yet downgraded missing goto targets and macro call defs to a warning, in case they are in extensions.conf; I rectified this problem. Also, A goto in a macro to a target in a catch block was not being found; I fixed this too; the cause was that I needed to treat catch statements like an extension in the find_match code.
Russell Bryant [Tue, 27 Feb 2007 17:36:09 +0000 (17:36 +0000)]
Fix voicemail email attachments. I missed the conversion of one of the line
endings and there was an extra one where it should not have been.
(issue #9128)
Russell Bryant [Mon, 26 Feb 2007 20:42:21 +0000 (20:42 +0000)]
Restore the behavior of Asterisk 1.2 where if a device was not specified in
alsa.conf, then we just use the system default, instead of creating our own
default of hw:0,0. (issue #9139)
Joshua Colp [Mon, 26 Feb 2007 17:09:53 +0000 (17:09 +0000)]
Use ast_strlen_zero to see if the language and/or context argument is not present for Background instead of just checking if it is NULL. (issue #9141 reported by mjagdis)
Russell Bryant [Mon, 26 Feb 2007 00:35:04 +0000 (00:35 +0000)]
Blocked revisions 56729 via svnmerge
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r56729 | russell | 2007-02-25 18:34:31 -0600 (Sun, 25 Feb 2007) | 4 lines
Ensure that lock.h is included in utils.c with AST_API_MODULE defined so that
the implementations will be properly included when the AST_INLINE_API functions
are not going to be inlined. (issue #9124, festr)
projects / thirdparty / asterisk.git / log
