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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r49313 | crichter | 2007-01-03 10:06:50 +0100 (Mi, 03 Jan 2007) | 41 lines
Merged revisions 48319,48321,48467,48552,48576,49135,49303 via svnmerge from
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r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) | 1 line
changed a few debugs to higher debug levels
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r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) | 1 line
added the export and import of the MISDN_ADDRESS_COMPLETE Variable to inidcate wether the extension is already completely dialed or if there might come additional digits by information elements. also added some docs for that.
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r48467 | crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line
removed FIXUP state. added check for channel allocation conflict when we create a setup while the other site creates a setup on the same channel, besides the check we resolve this conflict.
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r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18 Dez 2006) | 1 line
when our PTP Partner sends us a SETUP with a preselected channel we just accept it, even when we're NT. added some checks for segfaults.
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r48576 | crichter | 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line
when we reject a channel, because it's in use already, we shouldn't process the setup anymore. made the channel allocation a bit easier and more understandable, removed a few unused lines
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r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02 Jan 2007) | 1 line
added check for channel ranges in the set/empty channel functions. set pmp_l1_check default to no. added misdn restart pid cli command. added cleaning of channel when we send a RELEASE_COMPLETE.
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r49303 | crichter | 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines
* Added check for bridging in misdn_call to avoid setting echocancellation
when 2 mISDN channels are involved and when bridging is set. That lead
to a kernel panic before under different situations, because we switched
about 2 times between hardware bridging and echocancelation
* readded MISDN_URATE variable which got lost before, this should make app_v110
work again
* fixed typo
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2. Rename 'minmessage' to 'minsecs' for parity.
3. Make 'maxsecs' a per-user option, in addition to global.
(Issue # 8624)
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this, implementing locking of this list to make it thread-safe.
- Add a "redirect" option to http.conf that allows redirecting one URI to
another. I was inspired to do this while playing with the Asterisk GUI. I
got tired of typing this URL to get to the GUI:
http://localhost:8088/asterisk/static/config/cfgadvanced.html
So, now I have the following line in http.conf:
redirect=/=/asterisk/static/config/cfgadvanced.html
Now, I can type the following into my browser and go to the GUI:
http://localhost:8088
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config files
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are passed as an argument.
- Update the code in main/http.c to use the new interface
(the diff is large but mostly mechanical, due to the name change of
several variables);
- And since now it is trivial, implement "AMI over TLS", and document
the possible options in manager.conf
- And since the test client (openssl s_client -connect host:port )
does not generate \r\n as a line terminator, make get_input()
also accept just a \n as a line terminator (Mac users: do you
also need the \r-only version ?)
The option parsing in manager.conf is not very efficient, and needs
to be cleaned up and made similar to what we have in http.conf
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48323 | russell | 2006-12-06 11:15:45 -0500 (Wed, 06 Dec 2006) | 11 lines
Merged revisions 48322 via svnmerge from
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r48322 | russell | 2006-12-06 11:05:54 -0500 (Wed, 06 Dec 2006) | 3 lines
Fix the name of the rtignoreregexpire option in the sample configuration file.
(issue #8526, arkadia)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48230 | qwell | 2006-12-04 11:54:46 -0600 (Mon, 04 Dec 2006) | 4 lines
Add documentation to voicemail.conf.sample for ODBC storage.
Issue 8499 - patch by blitzrage.
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- Encapsulate RTP timers to the RTP structure, so we have one set for video and one for audio
- Document RTP keepalive configuration option
- Cleanup and document the monitor support function to hangup on RTP timeouts
- Add RTP keepalive to SIP show settings
Imported from 1.4 with modifications for trunk.
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r48186 | qwell | 2006-12-01 14:25:51 -0600 (Fri, 01 Dec 2006) | 10 lines
Merged revisions 48183 via svnmerge from
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r48183 | qwell | 2006-12-01 14:19:10 -0600 (Fri, 01 Dec 2006) | 2 lines
Fix a small typo - issue 8848, reported by pabelanger
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(imported from 1.4)
- Small fixes to limitonpeer
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r48143 | file | 2006-11-30 12:57:35 -0500 (Thu, 30 Nov 2006) | 10 lines
Merged revisions 48142 via svnmerge from
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r48142 | file | 2006-11-30 12:55:23 -0500 (Thu, 30 Nov 2006) | 2 lines
Document 'port' for SIP peers, came up because of the current mailing list thread. (issue #8450 reported by blitzrage)
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Because https is more secure than http, it usually
makes sense to keep this service more open than the
one on the unencrypted port.
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password was,
while also still changing the password "internally".
Issue 7371, initial patch by pdunkel, with rewrite/config comments by me.
Additional modifications (yay bitmask) by pdunkel.
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Issue #8369, patch by ssuehring, modified slightly by me.
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for realtime users.
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- Add docs on canreinvite
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like 'strict' except it does not count paused queue members as unavailable. (issue #8263 reported by gnarf)
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6453)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47279 | kpfleming | 2006-11-07 12:56:21 -0600 (Tue, 07 Nov 2006) | 2 lines
clean up sample config, and make native file playback the more obvious default choice
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together and comment them out by default.
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to the peer side of a type=friend.
This is for trying to support BJ in his quest to solve some issues
with the queue system and type=friend objects.
BJ: Please test!
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manager list -> manager show
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...where did the 1.2 properties come from, really? they're back.
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is to the person that configures asterisk. That we use it internally in the
contact header is a totally different story.
Still not convinced this is a good option.
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see queues.conf.sample for details.
* Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
setqueueentryvar options for each queue, see queues.conf.sample for details.
(#8216, jmls reported and submitted)
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application by configuring them in voicemail.conf (issue #7415, patch by
fkasumovic, with some fixes and documentation updates by myself)
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r46351 | crichter | 2006-10-27 11:49:20 +0200 (Fr, 27 Okt 2006) | 9 lines
Merged revisions 46176 via svnmerge from
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r46176 | crichter | 2006-10-25 10:41:59 +0200 (Mi, 25 Okt 2006) | 1 line
added nttimeout option to configure wether we disconnect calls on NT timeouts or not during an overlapdial session
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r46352 | crichter | 2006-10-27 11:58:44 +0200 (Fr, 27 Okt 2006) | 1 line
fixed not compile issue, which was just introduced
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r46353 | crichter | 2006-10-27 12:03:23 +0200 (Fr, 27 Okt 2006) | 9 lines
Merged revisions 46350 via svnmerge from
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r46350 | crichter | 2006-10-27 11:24:01 +0200 (Fr, 27 Okt 2006) | 1 line
fixed a bug which caused chan_misdn to try to allocate 2 times the same channel on high load, which then caused instability of mISDN. removed a useless function from isdn_lib.c
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when the number of channels fill the MTU on a given link.
In the future, this needs to be configurable per peer with trunking enabled.
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and support linux as well (using fopencookie(), which should
be available in glibc).
Update configure.ac to check for funopen (BSD) and fopencookie(glibc),
and while we are at it also for gethostbyname_r
(the generated files need to be updated, or you need
to run bootstrap.sh yourself).
Document the new options in http.conf.sample
(names are only tentative, better ones are welcome).
At this point we can safely enable the option.
Anyone willing to try this on Sun and Apple platforms ?
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r45439 | russell | 2006-10-17 22:19:07 -0400 (Tue, 17 Oct 2006) | 2 lines
update entry to reboot a snom phone (issue #7850, pnlarsson)
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replies, then ignore responses and finally I'm just getting rid of the option altogether and making it the default no matter what. C'est la vie!
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r45280 | file | 2006-10-16 16:06:18 -0400 (Mon, 16 Oct 2006) | 10 lines
Merged revisions 45265 via svnmerge from
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r45265 | file | 2006-10-16 15:59:54 -0400 (Mon, 16 Oct 2006) | 2 lines
Use responses rather then replies even though they mean the same thing.
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r45262 | file | 2006-10-16 15:37:34 -0400 (Mon, 16 Oct 2006) | 10 lines
Merged revisions 45260 via svnmerge from
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r45260 | file | 2006-10-16 15:28:18 -0400 (Mon, 16 Oct 2006) | 2 lines
Add 'ignoreoodreplies' option which will not create a pvt structure on a SIP response but instead basically drop it.
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