Add and extend the see-also sections to the documentation for applications
and functions in an effort to expand the online documentation of the wiki.
Also check for and update any links to moved documentation in the doc folder.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@304908 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r302548 | seanbright | 2011-01-19 13:37:09 -0500 (Wed, 19 Jan 2011) | 10 lines
Properly handle partial reads from fgets() when handling AGIs.
When fgets() failed with EAGAIN, we were continually decrementing the available
space left in our buffer, resulting in botched command handling.
(closes issue #16032)
Reported by: notahat
Patches:
agi_buffer_patch2.diff uploaded by fnordian (license 110)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@302549 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r300622 | tilghman | 2011-01-05 12:54:58 -0600 (Wed, 05 Jan 2011) | 17 lines
Merged revisions 300621 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r300621 | tilghman | 2011-01-05 12:47:46 -0600 (Wed, 05 Jan 2011) | 10 lines
Use the sanity check in place of the disconnect/connect cycle.
The disconnect/connect cycle has the potential to cause random crashes.
(closes issue #18243)
Reported by: ks3
Patches:
res_odbc.patch uploaded by ks3 (license 1147)
Tested by: ks3
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r299448 | tilghman | 2010-12-22 14:03:30 -0600 (Wed, 22 Dec 2010) | 8 lines
Resolve warnings by disambiguating the "s" extension as used by chan_dahdi from the "s" extension as used by the AEL macros.
(closes issue #18480)
Reported by: nivek
Patches:
20101215__issue18480__2.diff.txt uploaded by tilghman (license 14)
Tested by: nivek
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@299449 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Tweak the way fax stats are calculated so that all fax attempts and faliures are logged. Also make ensure faxes are either counted as completed or falied and never both.
FAX-210
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@297905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r294277 | jpeeler | 2010-11-08 15:58:13 -0600 (Mon, 08 Nov 2010) | 16 lines
Fix playback failure when using IAX with the timerfd module.
To fix this issue the alert pipe will now be used when the timerfd module is
in use. There appeared to be a race that was not solved by adding locking in the
timerfd module, but needed to be there anyway. The race was between the timer
being put in non-continuous mode in ast_read on the channel thread and the IAX
frame scheduler queuing a frame which would enable continuous mode before the
non-continuous mode event was read. This race for now is simply avoided.
(closes issue #18110)
Reported by: tpanton
Tested by: tpanton
I put tested by tpanton because it was tested on his hardware. Thanks for the
remote access to debug this issue!
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294278 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds information about the encryption setting to 'sip show
peers' and removes an out-of-date comment from res_srtp.c and instead
directs users to the proper documentation.
(closes issue #18140)
Reported by: chodorenko
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292309 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r292049 | tzafrir | 2010-10-16 12:03:04 +0200 (ש', 16 אוק 2010) | 15 lines
Base directory for MOH should be ASTDATADIR
If the directive 'directory' is relative, make it relative to the
datadir, rather than to the varlibdir. In the sample configuration
it is relative ('moh').
This has no effect unless you have actively set the datadir explicitly
(at build time or at run time).
(closes issue #16906)
Patches:
moh_datadir uploaded by tzafrir (license 46)
Review: https://reviewboard.asterisk.org/r/974/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This avoids unhappy crashing when we try to 'core stop gracefully' and res_srtp
tries to unload before chan_sip does. Thanks, Russell!
(closes issue #18085)
Reported by: st
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@292016 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r291904 | twilson | 2010-10-15 09:16:57 -0700 (Fri, 15 Oct 2010) | 7 lines
Don't crash or deadlock on module unload
We can't hold the lock while pthread_join is called since aji_log_hook will
attempt to lock from the other therad. We reorder the pthread_join and
ast_aji_disconnect so that we don't do an SSL_read() while SSL_shutdown is
running, causing a crash.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch includes several chan_gtalk enhancements.
Two new gtalk.conf options have been added, externip
and stunadd. Setting externip allows us to
manually specify what the external IP address is
outside of a NAT environment. Setting the stunaddr
option to a valid stun server allows for that external
ip to be retrieved via a STUN server automatically. This
external IP is then advertised during call setup as
a possible candidate.
I have also attempted to clean up chan_gtalk's code
so it meets our coding guidelines. During this cleanup
I noticed several things that need to be done in the
code and made a TODO section at the top of the file.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@291192 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It is possible for ast_rtp_stop() to be called which will clear the remote
address and cause the sendto to fail and spam warnings. Don't send in this
case.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@290542 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r290254 | tilghman | 2010-10-04 18:14:59 -0500 (Mon, 04 Oct 2010) | 11 lines
Change new pattern matcher to regard dashes the same as the old pattern matcher -- as visual candy to be ignored.
Also change the AEL parser to not generate dashes within extensions, as those
dashes would be ignored. Update the AEL tests to match this behavior.
(closes issue #17366)
Reported by: murf
Patches:
20100727__issue17366.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
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r289798 | jpeeler | 2010-10-01 18:01:31 -0500 (Fri, 01 Oct 2010) | 22 lines
Merged revisions 289797 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r289797 | jpeeler | 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010) | 15 lines
Change RFC2833 DTMF event duration on end to report actual elapsed time.
The scenario here is with a non P2P early media session. The reported time
length of DTMF presses are coming up short when sending to the remote side.
Currently the event duration is a running total that is incremented when sending
continuation packets. These continuation packets are only triggered upon
incoming media from the remote side, which means that the running total probably
is not going to end up matching the actual length of time Asterisk received
DTMF. This patch changes the end event duration to be lengthened if it is
detected that the end event is going to come up short.
Review: https://reviewboard.asterisk.org/r/957/
ABE-2476
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@289840 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Also make it more obvious when there is an issue en/decrypting.
(closes issue #17563)
Reported by: Alexcr
Patches:
res_srtp.c.patch uploaded by sfritsch (license 1089)
Tested by: twilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@287056 65c4cc65-6c06-0410-ace0-fbb531ad65f3
