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Author SHA1 Message Date
oej d0bcdac211 Manager events from the "moremanager" branch
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89706 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27 19:50:12 +00:00
kpfleming 094b009478 Merged revisions 89701 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89701 | kpfleming | 2007-11-27 13:36:55 -0600 (Tue, 27 Nov 2007) | 2 lines

generate a warning when an application option that requires an argument is ignored due to lack of an argument

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89704 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27 19:47:19 +00:00
oej 13d6371f2b Starting to merge changes from the "moremanager" branch. Documentation will
follow.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89702 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27 19:45:39 +00:00
oej bb9210c7a9 The following patch with updates for trunk. Works much better in trunk.
Also by accident fixed a bad typo by a previous committer, which actually made video calls
not work fully...

Merged revisions 89630 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89630 | oej | 2007-11-27 16:23:17 +0100 (Tis, 27 Nov 2007) | 12 lines

If we get a codec offer using a well-known payload type, but using it for another
codec that we don't know, Asterisk did not remove that codec from the list.

With this patch, we remove the codec from audio and video rtp objects and
deny it ever existed. Thanks to lasse for testing.

(closes issue #11376)
Reported by: lasse
Patches: 
      bug11376.txt uploaded by oej (license 306)
Tested by: lasse

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89698 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27 19:24:17 +00:00
qwell 3db53911ba Add an S_COR macro, which is similar to the existing S_OR macro,
except with an additional boolean arg.

A hack such as:
foo ? S_OR(bar, "baz") : "baz"
becomes:
S_COR(foo, bar, "baz")


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89683 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27 19:12:33 +00:00
murf 0c96bd8c5f made AEL 8-bit transparent; mainly the lexer was tossing chars with the hi-order bit set. Not nice. Also, allow @ in extension names, and a backslash, also.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89682 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27 18:50:44 +00:00
file 942e20012a Ensure the value returned from ast_random is between 0 and RAND_MAX on 64-bit platforms.
(closes issue #11348)
Reported by: sperreault


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89637 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27 17:01:19 +00:00
russell ba864b3835 Merged revisions 89634 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89634 | russell | 2007-11-27 10:12:33 -0600 (Tue, 27 Nov 2007) | 3 lines

Add a note to the sample voicemail config noting that when using IMAP storage,
only the first format specified will be attached to the message.

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89635 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27 16:13:14 +00:00
tilghman 2135451113 Merged revisions 89631 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89631 | tilghman | 2007-11-27 09:38:03 -0600 (Tue, 27 Nov 2007) | 3 lines

Default result of STAT should be "0" not "".
Reported via the -users mailing list, fixed by me.

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89632 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27 15:41:46 +00:00
oej 4c27a322a0 Merged revisions 89624 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89624 | oej | 2007-11-27 08:34:19 +0100 (Tis, 27 Nov 2007) | 6 lines

Clarify limitonpeers=yes

(closes issue #11304)
Reported by: pj


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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89625 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27 07:36:54 +00:00
murf 5aff21b945 Merged revisions 89622 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89622 | murf | 2007-11-26 23:24:02 -0700 (Mon, 26 Nov 2007) | 1 line

closes issue #11379; OK, this is an attempt to make both sides happy. To the cdr.conf file, I added the option 'unanswered', which defaults to 'no'. In this mode, you will see a cdr for a call, whether it was answered or not. The disposition will be NO ANSWER or ANSWERED, as appropriate. The src is as you'd expect, the destination channel will be one of the channels from the Dial() call, usually the last in the list if more than one chan was specified. With unanswered set to 'yes', you will still see this cdr entry in both cases. But in the case where the dial timed out, you will also see a cdr for each line attempted, marked NO ANSWER, with no destination channel name. The new option defaults to 'no', so you don't see the pesky extra cdr's by default, and you will not see the irritating 'not posted' messages.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89623 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27 06:47:08 +00:00
mmichelson 3d682ee9e4 Change all instances of "CALLERID(number)" to "CALLERID(num)" for
consistency's sake

(closes issue #11381, reported and patched by jon)



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89621 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 23:15:53 +00:00
mmichelson bbc2a86e79 Merged revisions 89618 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89618 | mmichelson | 2007-11-26 17:10:49 -0600 (Mon, 26 Nov 2007) | 7 lines

After issuing a "say load new", if a caller hangs up during the middle of playback of a number,
app_playback will continue to try to play the remaining files. With this change, no more files will
be played back upon hangup.

(closes issue #11345, reported and patched by IgorG)


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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89619 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 23:11:29 +00:00
mmichelson f0367e88b7 Blocked revisions 89616 via svnmerge
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r89616 | mmichelson | 2007-11-26 17:02:30 -0600 (Mon, 26 Nov 2007) | 5 lines

After issuing a "say load new" tons of warning messages are printed
out to the CLI every time do_say in app_playback is called. Removing these
warnings


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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89617 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 23:05:12 +00:00
russell 015c16d770 Update the configure script check for libpri to check for the newest function
that was just added.

Cresl1n, please keep this in mind when making these changes to libpri or libss7.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89615 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 22:52:36 +00:00
oej 4fd45884a2 Rename "limitonpeer" to "counteronpeer" since the call-limit is deprecated.
Both still works in this version.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89613 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 21:23:48 +00:00
file db19ec718e Merged revisions 89610 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89610 | file | 2007-11-26 17:10:29 -0400 (Mon, 26 Nov 2007) | 2 lines

Fix issues with async dialing with an application executing. The application has to be terminated and control returned to the thread before hanging things up. (issue #BE-252)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89612 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 21:14:07 +00:00
oej 4e62295db7 Formatting, doxygenification
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89611 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 21:12:50 +00:00
oej 812477c1c5 Formatting changes, cleaning up some code
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89609 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 20:55:09 +00:00
oej a6d5c8a789 Start using Doxygen groupings to group variables and defines.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89607 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 20:19:50 +00:00
oej d33873fade - Mark "concise" as deprecated
- Restructure other changes to UPGRADE.txt and CHANGES

We're still looking for scripts that replace 
	asterisk -rx "show shannels concise"
by using the manager interface, but still produces the same output.
Anyone?


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89606 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 19:24:23 +00:00
file 78ee740bbf Perform some module use counting audits. This is now done outside the scope of the application/dialplan function so they do not need to worry about it.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89602 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 18:11:31 +00:00
file b2469bcae6 Merged revisions 89599 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89599 | file | 2007-11-26 14:02:56 -0400 (Mon, 26 Nov 2007) | 6 lines

Add module counting removal for error conditions.
(closes issue #11333)
Reported by: Laureano
Patches:
      res_features_v2.c.patch uploaded by Laureano (license 265)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89600 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 18:04:46 +00:00
russell 4d2ea03db9 Merged revisions 89594 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89594 | russell | 2007-11-26 11:41:04 -0600 (Mon, 26 Nov 2007) | 3 lines

Add channel locking to a function that needed to be doing it.  This is just a
little something I noticed while working on a completely unrelated issue.

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89596 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 17:49:47 +00:00
murf 753634f9ca closes issue #11341; made changes to make utils again right with the MTX_PROFILE world.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89595 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 17:46:41 +00:00
file 6a8e5f1e64 Merged revisions 89592 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89592 | file | 2007-11-26 13:36:45 -0400 (Mon, 26 Nov 2007) | 6 lines

Use ast_free to free memory, or else we shall implode if MALLOC_DEBUG is enabled.
(closes issue #11347)
Reported by: ys
Patches:
      pbx.pbx_config.c.diff uploaded by ys (license 281)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89593 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 17:38:57 +00:00
murf bbd5a3d02e closes issue #11356; Many thanks to snuffy for his code review and changes to cut down duplication. I tested this against hashtest, and it passes. I reviewed the changes, and they look reasonable. I had to remove a few const decls to make things compile on my workstation,
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89591 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 17:26:01 +00:00
russell e09e950389 make sure we check to see if the configure script has been executed on a new checkout or after a distclean
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89590 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 17:25:08 +00:00
file c585cac92c Merged revisions 89587 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89587 | file | 2007-11-26 13:20:58 -0400 (Mon, 26 Nov 2007) | 6 lines

Close the audio file before sending it to the post processing application.
(closes issue #11357)
Reported by: reformed
Patches:
      mixmonitor.patch uploaded by reformed (license 330)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89589 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 17:23:28 +00:00
kpfleming 9af283654a Merged revisions 89586 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89586 | kpfleming | 2007-11-26 11:20:36 -0600 (Mon, 26 Nov 2007) | 2 lines

when parsing application options that take arguments, don't indicate that the option was supplied unless a non-zero-length argument was found for it

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89588 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 17:21:37 +00:00
murf 4f8e82fa2b Thanks to pnlarsson for noting the spelling error in the cli commands. Also, added some verbage about the new algorithm to CHANGES.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89583 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 16:24:27 +00:00
file 60d6d5e6e4 Revert change for 11348 until it can be looked at even more.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89582 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 16:20:04 +00:00
mmichelson 43aebe686f Merged revisions 89580 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89580 | mmichelson | 2007-11-26 09:48:06 -0600 (Mon, 26 Nov 2007) | 6 lines

Revert vmu->email back to an empty string if it was empty when imap_store_file
was called. This prevents sending a duplicate e-mail. 

(closes issue #11204, reported by spditner, patched by me)


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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89581 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 15:50:37 +00:00
file 532589d879 Merged revisions 89577 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89577 | file | 2007-11-26 11:34:38 -0400 (Mon, 26 Nov 2007) | 6 lines

If channel allocation fails because the alert pipe could not be created also free the scheduler context.
(closes issue #11355)
Reported by: eliel
Patches:
      main.channel.c.patch uploaded by eliel (license 64)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89578 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 15:36:27 +00:00
file 5219dc885e Make the behavior of using /dev/urandom for random numbers the same as random().
(closes issue #11348)
Reported by: sperreault
Patches:
      ast_random2.diff uploaded by sperreault (license 252)


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89576 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 15:28:13 +00:00
file d7b36511c0 Instead of printing out one codec in sip show channels print out all of the native ones (this is for video).
(closes issue #11366)
Reported by: ovi


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89573 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 14:50:51 +00:00
file 8ad982f1ba Merged revisions 89571 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89571 | file | 2007-11-26 10:41:03 -0400 (Mon, 26 Nov 2007) | 4 lines

When unloading app_meetme destroy any auto created contexts created by SLA.
(closes issue #11367)
Reported by: eliel

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89572 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 14:42:57 +00:00
file b8f8021a8a Don't crash if the 'o' option of ControlPlayback is used without any value.
(closes issue #11375)
Reported by: johan


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89570 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 14:31:32 +00:00
oej 18ff1ee386 Formatting changes
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89566 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-25 21:12:25 +00:00
oej bc21278887 Try to get channel.h and channel.c aligned in regards to ast_set_callerid as well
as change name of variables to follow the rest of the naming.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89564 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-25 19:33:33 +00:00
tilghman 6ab028735f Merged revisions 89559 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89559 | tilghman | 2007-11-25 11:17:10 -0600 (Sun, 25 Nov 2007) | 14 lines

We previously attempted to use the ESCAPE clause to set the escape delimiter to
a backslash.  Unfortunately, this does not universally work on all databases,
since on databases which natively use the backslash as a delimiter, the
backslash itself needs to be delimited, but on other databases that have no
delimiter, backslashing the backslash causes an error.

So the only solution that I can come up with is to create an option in res_odbc
that explicitly specifies whether or not backslash is a native delimiter.  If
it is, we use it natively; if not, we use the ESCAPE clause to make it one.

Reported by: elguero
Patch by: tilghman
(Closes issue #11364)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89561 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-25 17:50:07 +00:00
tilghman 1f7c33b062 Typo (someone needs to test compile before committing his changes)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89560 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-25 17:44:16 +00:00
oej 88fbfcf126 More doxygen changes
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89557 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-25 12:18:35 +00:00
oej 6ee0d13116 Housekeeping
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89556 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-25 12:12:00 +00:00
oej fb88a42abe Formatting, doxygen updates
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89555 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-25 12:06:57 +00:00
oej 003485a22b - Deprecate "call-limit" in chan_sip. No other channel driver enforces call-limits
and we now have the groupcount system to implement call-limits in the dialplan. You
  can use the "setvar" option in realtime/sip.conf to set limits per device.

- Implement "callcounter" as a new option to enable the call counting we need to
  report device status to queue, manager and SIP subscriptions.

The call counter setting is now enabled in the code by setting the device call-limit
to 999. When we remove the call limit, we can simply enable this with a boolean
setting.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89554 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-25 11:46:17 +00:00
oej 14c325e930 Housekeeping...
- Fix typo in chan_sip
- Remove changes to caller ID structure, moving it to branch (russellb)


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89551 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-25 11:10:52 +00:00
murf 2a36d53ce3 closes issue #11363; where the pattern _20x. buried in an included context, didn't match 2012; There were a small set of problems to fix: 1. I needed NOT to score patterns unless you are at the end of the data string. 2. Capital N,X,Z and small n,x,z are OK in patterns. I canonicalize the patterns in the trie to caps. 3. When a pattern ends with dot or exclamation, CANMATCH/MATCHMORE should always report this pattern, no matter the length. With this commit, I also supplied the wish of Luigi, where the user can select which pattern matching algorithm to use, the old (legacy) pattern matcher, or the new, trie based matcher. The OLD matcher is the default. A new [general] section variable, extenpatternmatchnew, is added to the extensions.conf, and the example config has it set to false. If true, the new matcher is used. In all other respects, the context/exten structs are the same; the tries and hashtabs are formed, but in the new mode the tries are not used. A new CLI command 'dialplan set extenpatternmatch true/false' is provided to allow switching at run time. I beg users that are forced to return to the old matcher to please report the reason in the bug tracker. Measured the speed benefit of the new matcher against an impossibly large context with 10,000 extensions: the new matcher is 374 times faster.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89547 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-24 21:00:26 +00:00
tilghman 2794d1de7a Merged revisions 89545 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89545 | tilghman | 2007-11-24 10:59:59 -0600 (Sat, 24 Nov 2007) | 5 lines

Free some frames that would otherwise leak on error.
Reported by: Laureano
Patch by: Laureano,tilghman
(Closes issue #11351)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89546 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-24 17:07:12 +00:00
murf f0138f250b Added <sys/file.h> include to allow trunk to compile. Hope this doesn't louse thing up.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89544 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-24 16:53:24 +00:00