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configure script command line option for OpenAIS
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@127017 f38db490-d61c-443f-a65b-d21fe96a405b
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by Noel Morais on the users mailing list
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@125703 f38db490-d61c-443f-a65b-d21fe96a405b
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- change ast_settimeout() to honor max rate in edge cases of file playback
(this will make some warning messages go away at the end of playing back
a file)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@125332 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r125132 | kpfleming | 2008-06-25 17:21:30 -0500 (Wed, 25 Jun 2008) | 10 lines
allow tonezone to live in a different place than DAHDI/Zaptel, since dahdi-tools and dahdi-linux are now separate packages and can be installed in different places
don't include tonezone.h in dahdi_compat.h, because only a couple of modules need it
get app_rpt building again after the DAHDI changes
(closes issue #12911)
Reported by: tzafrir
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module.
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the corresponding roster item has a subscription value set to "none"
or "from".
Make the code more readable.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@124872 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@124870 f38db490-d61c-443f-a65b-d21fe96a405b
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(inspired by, and potentially fixes issue #12917)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@124798 f38db490-d61c-443f-a65b-d21fe96a405b
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can't find a record.
(Closes issue #12908)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@124505 f38db490-d61c-443f-a65b-d21fe96a405b
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They want (char *)NULL as sentinel.
An example is OpenBSD (confirmed on 4.3) that ships with gcc 3.3.4
This commit introduces a contstant SENTINEL which is declared as:
#define SENTINEL ((char *)NULL)
All places I could test compile on my openbsd system are converted.
Update CODING-GUIDELINES to tell about this constant.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@124127 f38db490-d61c-443f-a65b-d21fe96a405b
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- change the "timing test" CLI command to let you specify a timing rate to test
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@124023 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #12572)
Reported by: flyn
Patches:
20080613__bug12572.diff.txt uploaded by Corydon76 (license 14)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@123952 f38db490-d61c-443f-a65b-d21fe96a405b
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does not require DAHDI. It's called "pthread" because it uses a pthread
API call in the timing thread for sleeping and ensuring we wake up at
an appropriate time. I wasn't sure what else to call it. :)
The timing API requires a file descriptor that can be polled on. So,
when you open a timer, this module creates a pipe and returns the read
end of the pipe. There is a background thread that wakes up every 10ms
and checks to see if any of the currently open timers need a 'tick' and
writes to the appropriate pipe.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122928 f38db490-d61c-443f-a65b-d21fe96a405b
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add automatic use count handling for timing modules.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122526 f38db490-d61c-443f-a65b-d21fe96a405b
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- Convert chan_iax2 to use the timing API
- Convert usage of timing in the core to use the timing API instead of
using DAHDI directly
- Make a change to the timing API to add the set_rate() function
- change the timing core to use a rwlock
- merge a timing implementation, res_timing_dahdi
Basic testing was successful using res_timing_dahdi
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122523 f38db490-d61c-443f-a65b-d21fe96a405b
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Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122234 f38db490-d61c-443f-a65b-d21fe96a405b
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places (app_voicemail, for example).
(Related to bug #11678)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@121683 f38db490-d61c-443f-a65b-d21fe96a405b
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This commit merges in the rest of the code needed to support distributed device
state. There are two main parts to this commit.
Core changes:
- The device state handling in the core has been updated to understand device
state across a cluster of Asterisk servers. Every time the state of a device
changes, it looks at all of the device states on each node, and determines the
aggregate device state. That resulting device state is what is provided to
modules in Asterisk that take actions based on the state of a device.
New module, res_ais:
- A module has been written to facilitate the communication of events between
nodes in a cluster of Asterisk servers. This module uses the SAForum AIS
(Service Availability Forum Application Interface Specification) CLM and EVT
services (Cluster Management and Event) to handle this task. This module
currently supports sharing Voicemail MWI (Message Waiting Indication) and
device state events between servers. It has been tested with openais, though
other implementations of the spec do exist.
For more information on testing distributed device state, see the following doc:
- doc/distributed_devstate.txt
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to realtime less painful in the future.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@120789 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r120675 | phsultan | 2008-06-05 18:56:15 +0200 (Thu, 05 Jun 2008) | 2 lines
Ignore appended resource when comparing JIDs.
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has been loaded, fix a return value in the loader, and ensure that the help
workhorse header does not print on load.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@120602 f38db490-d61c-443f-a65b-d21fe96a405b
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and off for new installations. This includes the translation from pipes to commas
for pbx_realtime and the EXEC command for AGI, as well as the change to the Set
application not to support multiple variables at once.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@120171 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r119929 | murf | 2008-06-03 08:49:46 -0600 (Tue, 03 Jun 2008) | 16 lines
as per http://lists.digium.com/pipermail/asterisk-users/2008-June/212934.html,
which is a message from Philipp Kempgen, requesting that the WARNING
that an extension is empty be reduced to a NOTICE or less, as empty
extensions are syntactically possible, and no big deal.
With which I agree, and have removed that WARNING message entirely.
I think it is not necessary to see this message. It didn't
state that a NoOp() was inserted automatically on your behalf,
and really, as users, who cares? Why freak out dialplan writers
with unnecessary warnings? The details of the machinations a compiler goes
thru to produce working assembly code is of little interest
to most programmers-- we will follow the unix principal of
doing our work silently.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@119930 f38db490-d61c-443f-a65b-d21fe96a405b
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jabber.conf). The actual connection is made when a call comes in
Asterisk.
Apply this fix to Jingle too.
Fix the ast_aji_get_client function that was not able to retrieve an
XMPP client from its JID.
(closes issue #12085)
Reported by: junky
Tested by: phsultan
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@119741 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #12601)
Reported by: suretec
Patches:
res_config_ldap.c.patch uploaded by suretec (license 70)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118302 f38db490-d61c-443f-a65b-d21fe96a405b
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(Closes issue #12704)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118129 f38db490-d61c-443f-a65b-d21fe96a405b
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them to the parser ;
- report Gtalk error messages from a buddy to the console.
This patch makes Asterisk "Google Jingle" (chan_gtalk) implementation
work with Empathy. Note that this is only true for audio streams, not
video.
Thank you to PH for his great help!
(closes issue #12647)
Reported by: PH
Patches:
trunk-12647-1.diff uploaded by phsultan (license 73)
Tested by: phsultan, PH
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118020 f38db490-d61c-443f-a65b-d21fe96a405b
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- make data member of the ast_frame struct a named union instead of a void
Recently the ast_queue_hangup function got a new parameter, the hangupcause
Feedback came in that this is no good and that instead a new function should be created.
This I did.
The hangupcause was stored in the seqno member of the ast_frame struct. This is not very
elegant, and since there's already a data member that one should be used.
Problem is, this member was a void *.
Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone
wants to store another type in there in the future.
This commit is so massive, because all ast_frame.data uses have to be
altered to ast_frame.data.data
Thanks russellb and kpfleming for the feedback.
(closes issue #12674)
Reported by: mvanbaak
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117802 f38db490-d61c-443f-a65b-d21fe96a405b
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(Related to issue #12677)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117264 f38db490-d61c-443f-a65b-d21fe96a405b
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destruction, for future users.
(Closes issue #12677)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117262 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r117135 | file | 2008-05-19 13:50:52 -0300 (Mon, 19 May 2008) | 6 lines
Use the right pthread lock and condition when waiting.
(closes issue #12664)
Reported by: tomo1657
Patches:
res_smdi.c.patch uploaded by tomo1657 (license 484)
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destroying it) and use a callback to purge remaining classes.
(closes issue #12677)
Reported by: falves11
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117133 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r116466 | tilghman | 2008-05-14 16:38:09 -0500 (Wed, 14 May 2008) | 7 lines
Avoid zombies when the channel exits before the AGI.
(closes issue #12648)
Reported by: gkloepfer
Patches:
20080514__bug12648.diff.txt uploaded by Corydon76 (license 14)
Tested by: gkloepfer
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@116467 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #12613)
Reported by: suretec
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trunk.
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(closes issue #12600)
Reported By: suretec
Patch by me
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115523 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #12599)
Reported by: suretec
Patches:
12599.patch uploaded by juggie (license 24)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115521 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r115418 | qwell | 2008-05-06 14:34:58 -0500 (Tue, 06 May 2008) | 7 lines
Switch to using ast_random() rather than just rand().
This does not fix the bug reported, but I believe it is correct.
(from issue #12446)
Patches:
bug_12446.diff uploaded by snuffy (license 35)
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(closes issue #10540)
Reported by: spendergrass
Patches:
20080417__bug10540.diff.txt uploaded by Corydon76 (license 14)
Tested by: blitzrage
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115076 f38db490-d61c-443f-a65b-d21fe96a405b
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This commit adds some new features to the SMDI_MSG_RETRIEVE() dialplan function.
Previously, this function only allowed searching by the forwarding station.
I have added some options to allow you to also search for messages in the queue
by the message desk terminal ID, as well as the message desk number.
This originally came up as a suggestion on the asterisk-dev mailing list.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r114829 | qwell | 2008-04-29 12:08:55 -0500 (Tue, 29 Apr 2008) | 1 line
Change warning message to debug, since there are cases where 0 results is perfectly fine.
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