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r299449 | tilghman | 2010-12-22 14:05:02 -0600 (Wed, 22 Dec 2010) | 15 lines
Merged revisions 299448 via svnmerge from
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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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Previously, I had added the ast_sched_thread stuff that was a generic scheduler
thread implementation. However, if you used it, it required using different
functions for modifying scheduler contents. This patch reworks how this is
done and just allows you to optionally start a thread on the original scheduler
context structure that has always been there. This makes it trivial to switch
to the generic scheduler thread implementation without having to touch any of
the other code that adds or removes scheduler entries.
In passing, I made some naming tweaks to add ast_ prefixes where they were not
there before.
Review: https://reviewboard.asterisk.org/r/1007/
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r297157 | mnicholson | 2010-12-01 13:47:33 -0600 (Wed, 01 Dec 2010) | 2 lines
Changed some NOTICE and WARNING messages to DEBUG messages.
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r297486 | mnicholson | 2010-12-02 15:30:47 -0600 (Thu, 02 Dec 2010) | 6 lines
Add support for reserving a fax session before answering the channel.
Note: this change breaks ABI compatibility.
FAX-217
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r297495 | mnicholson | 2010-12-03 09:21:52 -0600 (Fri, 03 Dec 2010) | 4 lines
Print a DEBUG message instead of a WARNING message when the selected fax tech does not support reserving sessions.
Answer the channel before quering it for t.38 support. This is necessary for the query to work properly over local channels.
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Add the ability to play a sound file, listen for more than just one digit,
specify
escape characters. Backwards compatible (to work with only timeout specified).
(closes issue #15531)
Reported by: diLLec
Patches:
asterisk-res_agi-203638-patched.patch uploaded by diLLec (license 839)
Tested by: diLLec, espiceland
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r294278 | jpeeler | 2010-11-08 15:59:45 -0600 (Mon, 08 Nov 2010) | 23 lines
Merged revisions 294277 via svnmerge from
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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r292376 | tilghman | 2010-10-19 19:40:29 -0500 (Tue, 19 Oct 2010) | 5 lines
Oops. This module uses the generic timer and no longer uses DAHDI.
This causes a problem with the Solaris and other system builds that have gcc
4.1 (where optional_api is non-optional).
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r292309 | twilson | 2010-10-19 12:27:32 -0700 (Tue, 19 Oct 2010) | 10 lines
Add sip show peer info about crypto and remove dated comment
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
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r292050 | tzafrir | 2010-10-16 12:47:00 +0200 (ש', 16 אוק 2010) | 22 lines
Merged revisions 292049 via svnmerge from
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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r292016 | twilson | 2010-10-15 16:40:56 -0500 (Fri, 15 Oct 2010) | 5 lines
Ref/unref res_srtp when we create/destroy a session
This avoids unhappy crashing when we try to 'core stop gracefully' and res_srtp
tries to unload before chan_sip does. Thanks, Russell!
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r291905 | twilson | 2010-10-15 09:39:58 -0700 (Fri, 15 Oct 2010) | 14 lines
Merged revisions 291904 via svnmerge from
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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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r291192 | dvossel | 2010-10-11 16:38:39 -0500 (Mon, 11 Oct 2010) | 19 lines
Gtalk enhancements and general code cleanup.
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.
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r290542 | twilson | 2010-10-05 21:35:51 -0700 (Tue, 05 Oct 2010) | 6 lines
Don't try to send RTP when remote_address is null
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.
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r290255 | tilghman | 2010-10-04 18:23:11 -0500 (Mon, 04 Oct 2010) | 18 lines
Merged revisions 290254 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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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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r289840 | jpeeler | 2010-10-01 21:43:45 -0500 (Fri, 01 Oct 2010) | 29 lines
Merged revisions 289798 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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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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r287269 | pitel | 2010-09-17 10:37:49 +0200 (Pá, 17 zář 2010) | 8 lines
Support for HTTP redirects in calendar's URL
libneon does not support HTTP redirects (3xx responses) by default. You must tell it to follow them.
Also, another little unsigned int fix.
(closes issue #17776)
Review: https://reviewboard.asterisk.org/r/921/
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r287270 | pitel | 2010-09-17 10:42:37 +0200 (Pá, 17 zář 2010) | 6 lines
Asterisk crashing because of double free when EWS request fails
The free is done later in code. I think ast_free() should have built in checks for double free.
(closes issue #17782)
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r287271 | pitel | 2010-09-17 10:44:28 +0200 (Pá, 17 zář 2010) | 6 lines
Events are visible after they were removed from EWS calendar
Because we must merge calendar even when it's empty.
(closes issue #17786)
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r287056 | twilson | 2010-09-15 17:17:17 -0500 (Wed, 15 Sep 2010) | 10 lines
Don't hang up a call on an SRTP unprotect failure
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
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r284477 | twilson | 2010-09-01 13:44:36 -0500 (Wed, 01 Sep 2010) | 17 lines
Fix SRTP for changing SSRC and multiple a=crypto SDP lines
Adding code to Asterisk that changed the SSRC during bridges and masquerades
broke SRTP functionality. Also broken was handling the situation where an
incoming INVITE had more than one crypto offer. This patch caches the SRTP
policies the we use so that we can change the ssrc and inform libsrtp of the
new streams. It also uses the first acceptable a=crypto line from the incoming
INVITE.
(closes issue #17563)
Reported by: Alexcr
Patches:
srtp.diff uploaded by twilson (license 396)
Tested by: twilson
Review: https://reviewboard.asterisk.org/r/878/
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Jan Kalab
Tilghman Lesher
Russell Bryant
Matthew Nicholson
Brad Watkins
Erin Spiceland
Paul Belanger
Jeff Peeler
Terry Wilson
Tzafrir Cohen
David Vossel
Brett Bryant
Jason Parker
Leif Madsen