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r151464 | mmichelson | 2008-10-21 18:54:41 -0500 (Tue, 21 Oct 2008) | 11 lines
Make the sip_standard_port function more granular by allowing separate
type and port arguments. This is necessary because when building our From
and Contact headers, we need to be absolutely sure that we are placing our
source port there and not the peer's source port.
(closes issue #12761)
Reported by: asbestoshead
Patches:
patch-chan-sip-contact-port.txt uploaded by asbestoshead (license 455)
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r179469 | tilghman | 2009-03-02 17:10:18 -0600 (Mon, 02 Mar 2009) | 17 lines
Merged revisions 179468 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r179468 | tilghman | 2009-03-02 17:09:01 -0600 (Mon, 02 Mar 2009) | 10 lines
When ending a recording with silence detection, remember to reduce the duration.
The end of the recording is correspondingly trimmed, but the duration was not
trimmed by the number of seconds trimmed, so the saved duration was necessarily
longer than the actual soundfile duration.
(closes issue #14406)
Reported by: sasargen
Patches:
20090226__bug14406.diff.txt uploaded by tilghman (license 14)
Tested by: sasargen
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r179465 | russell | 2009-03-02 17:06:16 -0600 (Mon, 02 Mar 2009) | 4 lines
Fix a reference leak in timerfd_set_rate().
(found during a debugging session with dvossel and mmichelson.)
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r179462 | russell | 2009-03-02 17:00:30 -0600 (Mon, 02 Mar 2009) | 16 lines
Merged revisions 179461 via svnmerge from
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r179461 | russell | 2009-03-02 16:58:18 -0600 (Mon, 02 Mar 2009) | 8 lines
Ensure that only one thread is calling ast_settimeout() on a channel at a time.
For example, with an IAX2 channel, you can have both the channel thread and the
chan_iax2 processing threads calling this function, and doing so twice at the
same time is a bad thing.
(Found in a debugging session with dvossel and mmichelson)
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r179323 | file | 2009-03-02 10:28:09 -0400 (Mon, 02 Mar 2009) | 5 lines
Do not try to remove a registration scheduled item if the scheduler context has already been destroyed.
(closes issue #14580)
Reported by: alecdavis
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r179254 | mmichelson | 2009-03-01 17:25:23 -0600 (Sun, 01 Mar 2009) | 5 lines
Swap reversed timevals.
This was pointed out by ScribbleJ in #asterisk-dev. Thanks very much, ScribbleJ!
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This code already existed in trunk and 1.6.1, but was not in 1.6.0 prior to
this commit.
(closes issue #14338)
Reported by: fiddur
Patches:
14338.patch uploaded by mmichelson (license 60)
Tested by: fiddur
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r179219 | mmichelson | 2009-03-01 15:45:08 -0600 (Sun, 01 Mar 2009) | 18 lines
Properly free memory and remove scheduler entries when a transmission failure occurs.
Previously, only the "data" field of the sip_pkt created during __sip_reliable_xmit
was freed when XMIT_FAILURE was returned by __sip_xmit. When retrans_pkt was called,
this inevitably resulted in the reading and writing of freed memory.
XMIT_FAILURE is a condition meaning that we don't want to attempt resending the packet
at all. The proper action to take is to remove the scheduler entry we just created,
free the packet's data as well as the packet itself, and unlink it from the list of
packets on the sip_pvt structure.
(closes issue #14455)
Reported by: Nick_Lewis
Patches:
14455.patch uploaded by mmichelson (license 60)
Tested by: Nick_Lewis
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r179164 | russell | 2009-02-27 15:47:18 -0600 (Fri, 27 Feb 2009) | 2 lines
Mark res_ais as experimental, as the binary event format is subject to change.
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r179154 | russell | 2009-02-27 15:23:12 -0600 (Fri, 27 Feb 2009) | 2 lines
Add a note about the ordering of entries in sip.conf in 1.6.1.
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r179122 | mvanbaak | 2009-02-27 21:34:00 +0100 (Fri, 27 Feb 2009) | 16 lines
Add reload support to chan_skinny.
Special thanks goes to DEA who had to redo this patch twice
because we first put unload/load support in and later redid the way
we configure devices and lines.
(closes issue #10297)
Reported by: DEA
Patches:
skinny-reload-trunkv2.diff uploaded by wedhorn (license 30)
skinny-reload-trunk-v4.txt uploaded by DEA (license 3)
With mods by me based on feedback from wedhorn and Russell and seanbright
Tested by: DEA, mvanbaak, pj
Review: http://reviewboard.digium.com/r/130/
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r178986 | murf | 2009-02-26 20:45:58 -0700 (Thu, 26 Feb 2009) | 26 lines
Merged revisions 178956 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
In this case, it's just a matter of reducing the default timeouts from 2000
to 1000 msec, as the max def feature digit timeout is no longer halved.
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r178956 | murf | 2009-02-26 14:27:32 -0700 (Thu, 26 Feb 2009) | 18 lines
This change moves the default feature digit timeout to 1000 ms from the previous default of 500.
As per bug 14515, a dev discussion arrived at a "mediated concensus"
of a default feature digit timeout of 1.0 sec. Some voted for 1300;
ctooley thought 1500 for distracted phone users in phone booths;
kpfleming put his foot down at 1.0 sec.
Users who found the previous default max delay of 250 msec perfect,
are welcome to override the new default. Notice that I said that
250 msec was the default; wait a minute, you might say, the config
file said it was 500 msec!; well, because of the bug fix for 14515,
we found that 500 msec was actually enforcing a max of 250. The bug
fix would restore 500 msec, but we felt even that was a bit tight
for most users... 2000 msec was pushed earlier by mmichelson, so
that reduces to 1000 msec after the bug fix. Enjoy!
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r178870 | murf | 2009-02-26 10:45:22 -0700 (Thu, 26 Feb 2009) | 1 line
These small fixes prevent compiler warnings with ubuntu 8.10's gcc-4.3.2, which tend to break my dev-mode build. Not a problem in 1.6.x.
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r178828 | murf | 2009-02-26 10:22:11 -0700 (Thu, 26 Feb 2009) | 34 lines
Merged revisions 178804 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r178804 | murf | 2009-02-26 10:09:03 -0700 (Thu, 26 Feb 2009) | 28 lines
This patch prevents the feature detection timeout from being cut in half.
Because the ast_channel_bridge() call will return 0 and pass
a frame pointer for both DTMF_BEGIN and DTMF_END, the feature_timer
field in hte config struct is getting decremented twice, which
effectively cuts the digittimeout in half. I added conditions
to the if statement to only let DTMF_END frames to flow thru,
which solved the problem. Also, when the frame pointer is null,
let control flow thru-- this usually happens on timeouts. I added
a comment to the code to explain what's going on and why.
Many thanks to sodom for reporting this problem. Personnally, it always seemed
like something was wrong with the featuredigittimeout, but I never
could quite decide what... and was too busy to investigate.
This bug forced the issue, and now we know.
Sodom had other issues in 14515, but I couldn't reproduce them. If
he still has problems, and wants to get them solved, he is welcome
to reopen 14515.
(closes issue #14515)
Reported by: sodom
Patches:
14515.patch uploaded by murf (license 17)
Tested by: murf, sodom
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r178801 | file | 2009-02-26 12:42:36 -0400 (Thu, 26 Feb 2009) | 5 lines
Fix an issue where the timer for file playback would not be stopped if DAHDI was not installed.
(closes issue #14541)
Reported by: grant
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r178767 | dvossel | 2009-02-26 09:50:22 -0600 (Thu, 26 Feb 2009) | 8 lines
IAX2 prune realtime fix
Iax2 prune realtime had issues. If "iax2 prune realtime all" was called, it would appear like the command was successful, but in reality nothing happened. This is because the reload that was supposed to take place checks the config files, sees no changes, and does nothing. If there had been a change in the the config file, the realtime users would have been marked for deletion and everything would have been fine. Now prune_users() and prune_peers() are called instead of reload_config() to prune all users/peers that are realtime. These functions remove all users/peers with the rtfriend and delme flags set. iax2_prune_realtime() also lacked the code to properly delete a single friend. For example. if iax2 prune realtime <friend> was called, only the peer instance would be removed. The user would still remain.
(closes issue #14479)
Reported by: mousepad99
Review: http://reviewboard.digium.com/r/176/
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r178764 | file | 2009-02-26 11:40:10 -0400 (Thu, 26 Feb 2009) | 5 lines
Ensure there is a valid tone part before trying to play tones.
(closes issue #14558)
Reported by: alecdavis
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r178300 | dvossel | 2009-02-24 11:42:37 -0600 (Tue, 24 Feb 2009) | 14 lines
Allows manager command to see if IAX link is trunked and encrypted. Displays what kind of encryption is enabled as well.
Manager command "iaxpeers" now shows if a link is trunked and encrypted. Instead of encryption saying simply "yes" or "no", it now displays what type of encryption is enabled and if keyrotation is on or not.
(closes issue #14427)
Reported by: snuffy
Patches:
iax_show_trunks.diff uploaded by snuffy (license 35)
2009022200_iax2_show_trunkencryption.diff.txt uploaded by mvanbaak (license 7)
Tested by: mvanbaak, dvossel, snuffy
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r178213 | file | 2009-02-24 11:18:38 -0400 (Tue, 24 Feb 2009) | 16 lines
Merged revisions 178205 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r178205 | file | 2009-02-24 11:16:07 -0400 (Tue, 24 Feb 2009) | 9 lines
Skip check for extension when subscribing for MWI.
Since the remote side is not actually subscribing to a specific extension when
subscribing for MWI just skip the check to see if the extension exists. They can't use it
to specify the mailbox either since we require configuration of that in sip.conf
(closes issue #14531)
Reported by: festr
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r178142 | russell | 2009-02-23 17:11:37 -0600 (Mon, 23 Feb 2009) | 22 lines
Merged revisions 178141 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r178141 | russell | 2009-02-23 17:09:01 -0600 (Mon, 23 Feb 2009) | 14 lines
Fix infinite DTMF when a BEGIN is received without an END.
This commit is related to rev 175124 of 1.4 where a previous attempt was made
to fix this problem. The problem with the previous patch was that the inserted
code needed to go _before_ setting the lastrxts to the current timestamp.
Because those were the same, the dtmfcount variable was never decremented, and
so the END was never sent.
In passing, I removed the dtmfsamples variable which was completed unused. I
also removed a redundant setting of the lastrxts variable.
(closes issue #14460)
Reported by: moliveras
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r178030 | dvossel | 2009-02-23 11:59:55 -0600 (Mon, 23 Feb 2009) | 7 lines
Changes the way keyrotation is enabled by default
Key rotation was enabled by default by setting the global encryption method to IAX_ENCRYPT_KEYROTATE. the problem with this is that if encryption is not enabled, and the encryption method is set to anything except 0, the peer appears to have encryption enabled when issuing a "iax2 show peers". Rather than have the key rotation bit always set by default, it is now only set when an encryption method is enabled.
(closes issue #14523)
Reported by: mvanbaak
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r178061 | mvanbaak | 2009-02-23 19:23:38 +0100 (Mon, 23 Feb 2009) | 3 lines
update the new manager commands in chan_skinny to match
chan_sip's headers. requested by oej.
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r178027 | mvanbaak | 2009-02-23 18:48:32 +0100 (Mon, 23 Feb 2009) | 2 lines
list the addition of the SKINNY manager actions in the CHANGES file.
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r178022 | russell | 2009-02-23 11:29:16 -0600 (Mon, 23 Feb 2009) | 6 lines
Fix a regression in scheduler entry ordering, and add a regression test for it.
(closes issue #14522)
Reported by: pj
Tested by: russell
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r177852 | mvanbaak | 2009-02-21 14:13:35 +0100 (Sat, 21 Feb 2009) | 18 lines
set ASTVARRUNDIR=$(localstatedir)/run/asterisk as default path
When running asterisk as non-root and without this patch the pidfile wants
to go into /var/run/asterisk.pid. This directory is not writable for
the non-root user and changing permissions is not an option.
Putting it in /var/run/asterisk/asterisk.pid makes it possible
to set permissions on the /var/run/asterisk dir so everything
works as it should be.
Patched committed is based on pabelanger's patch.
(closes issue #13153)
Reported by: pabelanger
Patches:
2009012900_bug13153-nonrootscripts.diff.txt uploaded by mvanbaak (license 7)
Review: http://reviewboard.digium.com/r/139/
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r177787 | tilghman | 2009-02-20 17:02:35 -0600 (Fri, 20 Feb 2009) | 16 lines
Merged revisions 177786 via svnmerge from
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r177786 | tilghman | 2009-02-20 16:59:52 -0600 (Fri, 20 Feb 2009) | 9 lines
Don't print the CR-NL combination when we aren't outputting to the manager.
An embedded CR-NL in a CLI command screws up several AMI parsers that don't
expect to see that combination in the middle of output.
(Closes issue #14305)
Reported by: martins
Patch by: tilghman
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r177699 | dhubbard | 2009-02-20 14:29:00 -0600 (Fri, 20 Feb 2009) | 9 lines
Make app_fax compatible with spandsp-0.0.6pre4
Prior to spandsp-0.0.6pre4 the t30_stats_t structure used a pages_transferred
integer to indicate the number of pages transferred (so far) during the fax
session. The spandsp-0.0.6pre4 release removed the pages_transferred integer
and replaced it with two different integers - pages_tx and pages_rx. This
revision uses the new integers for spandsp-0.0.6pre4 while maintaining backwards
compatibility for previous spandsp releases.
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During iax2 call negotiation, supported codecs are passed in an Information Element containing a 2 byte field where each bit correlates to a specific codec. In 1.6 only audio codec bits 0-12 and 15 are defined, leaving bits 13-14 undefined. By default all bits are enabled unless specified otherwise. Since its a 2 byte field and 13-14 are not defined, these bits are never turned off. In trunk, bits 13-14 are defined, which means 1.6 is advertising support for codecs it does not have when talking to trunk. I fixed this by adding #define for undefined audio codec bits. These bits are then removed from iax2's full bandwidth capabilities.
(closes issue #14283)
Reported by: jcovert
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