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r162418 | russell | 2008-12-09 16:38:41 -0600 (Tue, 09 Dec 2008) | 7 lines
Add some additional Asterisk project developer documentation.
After the nightly update of the documentation on asterisk.org, I'll post
an update to asterisk-dev with a pointer to the changes. This covers some
release branch and commit policy information. None of this should be a
surprise, since it's just documenting what we have already been doing.
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r162414 | russell | 2008-12-09 16:25:06 -0600 (Tue, 09 Dec 2008) | 16 lines
Merged revisions 162413 via svnmerge from
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r162413 | russell | 2008-12-09 16:17:39 -0600 (Tue, 09 Dec 2008) | 8 lines
Remove the test_for_thread_safety() function completely.
The test is not valid. Besides, if we actually suspected that recursive
mutexes were not working, we would get a ton of LOG_ERROR messages when
DEBUG_THREADS is turned on.
(inspired by a discussion on the asterisk-dev list)
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r162291 | russell | 2008-12-09 14:59:54 -0600 (Tue, 09 Dec 2008) | 17 lines
Merged revisions 162286 via svnmerge from
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r162286 | russell | 2008-12-09 14:57:35 -0600 (Tue, 09 Dec 2008) | 9 lines
Fix an issue where callers on an incoming call on an SLA trunk would not hear ringback.
We need to make sure that we don't start writing audio to the trunk channel until we're
actually ready to answer it. Otherwise, the channel driver will treat it as inband
progress, even though all they are getting is silence.
(closes issue #12471)
Reported by: mthomasslo
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r161115 | dhubbard | 2008-12-04 17:00:30 -0600 (Thu, 04 Dec 2008) | 11 lines
If 'faxdetect=yes' in sip.conf, switch to a 'fax' extension (if it exists) after T38 is negotiated.
Terry Wilson created the original patch for this functionality, which I slightly modified and added
the faxdetect=yes|no configuration option. This patch is only for T38 fax detection and does not
do anything for G711 over SIP fax detection. By default, this option is disabled.
Reviewboard: http://reviewboard.digium.com/r/69/
This functionality is for issue AST-140.
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r162271 | murf | 2008-12-09 13:40:31 -0700 (Tue, 09 Dec 2008) | 9 lines
Merged revisions 162264 via svnmerge from
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r162264 | murf | 2008-12-09 13:20:54 -0700 (Tue, 09 Dec 2008) | 1 line
In discussion with seanbright on #asterisk-dev, I have added a default rule, and an option to suppress the default rule from being generated in the flex output, for the sake of those OS's where they didn't tweak flex's ECHO macro, and the compiler doesn't like it. The regressions are OK with this.
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r162266 | mmichelson | 2008-12-09 14:30:07 -0600 (Tue, 09 Dec 2008) | 14 lines
Merged revisions 162265 via svnmerge from
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r162265 | mmichelson | 2008-12-09 14:28:44 -0600 (Tue, 09 Dec 2008) | 6 lines
If we fail to start a thread for the pbx to run in, we need to
be sure to decrease the number of active calls on the system.
This fix may relate to ABE-1713, but it is not certain yet.
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r162205 | file | 2008-12-09 15:48:35 -0400 (Tue, 09 Dec 2008) | 14 lines
Merged revisions 162204 via svnmerge from
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r162204 | file | 2008-12-09 15:47:07 -0400 (Tue, 09 Dec 2008) | 7 lines
Make sure that the timestamp for DTMF is not the same as the previous voice frame and do not send audio when transmitting DTMF as this confuses some equipment.
(closes issue #13209)
Reported by: ip-rob
Patches:
13209.diff uploaded by file (license 11)
Tested by: ip-rob, bujones
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r162079 | murf | 2008-12-09 10:18:03 -0700 (Tue, 09 Dec 2008) | 53 lines
Merged revisions 162013 via svnmerge from
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r162013 | murf | 2008-12-09 09:31:55 -0700 (Tue, 09 Dec 2008) | 45 lines
(closes issue #14019)
Reported by: ckjohnsonme
Patches:
14019.diff uploaded by murf (license 17)
Tested by: ckjohnsonme, murf
This crash was the result of a few small errors that
would combine in 64-bit land to result in a crash.
32-bit land might have seen these combine to mysteriously
drop the args to an application call, in certain
circumstances.
Also, in trying to find this bug, I spotted
a situation in the flex input, where, in passing
back a 'word' to the parser, it would allocate
a buffer larger than necessary. I changed the
usage in such situations, so that strdup was
not used, but rather, an ast_malloc, followed
by ast_copy_string.
I removed a field from the pval struct, in
u2, that was never getting used, and set in
one spot in the code. I believe it was an
artifact of a previous fix to make switch
cases work invisibly with extens.
And, for goto's I removed a '!' from
before a strcmp, that has been there
since the initial merging of AEL2, that
might prevent the proper target of a
goto from being found. This was pretty
harmless on its own, as it would just
louse up a consistency check for users.
Many thanks to ckjohnsonme for providing
a simplified and complete set of information
about the bug, that helped considerably in
finding and fixing the problem.
Now, to get aelparse up and running again
in trunk, and out of its "horribly broken" state,
so I can run the regression suite!
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r161947 | eliel | 2008-12-09 12:49:30 -0200 (Tue, 09 Dec 2008) | 8 lines
Avoid allocating memory for a thread that don't need it. Also, this memory was not being freed until the
main thread ends. (That is never).
(closes issue #14040)
Reported by: eliel
Patches:
func_odbc.c.patch uploaded by eliel (license 64)
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r161951 | russell | 2008-12-09 08:57:39 -0600 (Tue, 09 Dec 2008) | 23 lines
Merged revisions 161948 via svnmerge from
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r161948 | russell | 2008-12-09 08:52:25 -0600 (Tue, 09 Dec 2008) | 15 lines
Fix a problem with GROUP() settings on a masquerade.
The previous code carried over group settings from the old channel to the new
one. However, it did nothing with the group settings that were already on the
new channel. This patch removes all group settings that already existed on the
new channel.
I have a more complicated version of this patch which addresses only the most
blatant problem with this, which is that a channel can end up with multiple
group settings in the same category. However, I could not think of a use case
for keeping any of the group settings from the old channel, so I went this route
for now.
(closes AST-152)
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r161637 | eliel | 2008-12-08 02:23:50 -0200 (Mon, 08 Dec 2008) | 4 lines
- Fix a leak while printing an argument description.
- Avoid printing the name of an argument in the [Arguments] tag if there is no description
for that argument.
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r161493 | mmichelson | 2008-12-05 17:24:38 -0600 (Fri, 05 Dec 2008) | 8 lines
If the autoloop flag is set on a channel, then we need to
add 1 to the priority when checking if the extension exists. Otherwise,
gosubs will fail.
This was discovered when investigating an asterisk-users mailing list post
made by Gary Hawkins.
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r161349 | seanbright | 2008-12-05 10:56:15 -0500 (Fri, 05 Dec 2008) | 5 lines
When using IMAP_STORAGE, it's important to convert bare newlines (\n) in
emailbody and pagerbody to CR-LF so that the IMAP server doesn't spit out an
error. This was informally reported on #asterisk-dev a few weeks ago. Reviewed
by Mark M. on IRC.
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r161350 | seanbright | 2008-12-05 11:04:36 -0500 (Fri, 05 Dec 2008) | 2 lines
Use ast_free() instead of free(), pointed out by eliel on IRC.
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r161181 | tilghman | 2008-12-04 23:41:41 -0600 (Thu, 04 Dec 2008) | 11 lines
The first file should have a blank config filename in the structure, so that
when a save occurs to a different filename, everything goes to the alternate
filename, instead of appending to the original. This is important for the
AMI command UpdateConfig.
(closes issue #13301)
Reported by: trevo
Patches:
20081113__bug13301.diff.txt uploaded by Corydon76 (license 14)
20081113__bug13301__1.6.0.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76, blitzrage
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r161014 | jpeeler | 2008-12-04 12:32:20 -0600 (Thu, 04 Dec 2008) | 17 lines
Merged revisions 161013 via svnmerge from
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r161013 | jpeeler | 2008-12-04 12:30:41 -0600 (Thu, 04 Dec 2008) | 9 lines
(closes issue #13835)
Reported by: matt_b
Tested by: jpeeler
This mirrors a check that was present in ast_rtp_read to also be in ast_rtp_raw_write to not schedule sending the receiver report if the remote RTCP endpoint address isn't present in the RTCP structure.
Closes AST-142.
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r160945 | mmichelson | 2008-12-04 10:45:06 -0600 (Thu, 04 Dec 2008) | 23 lines
Merged revisions 160943 via svnmerge from
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r160943 | mmichelson | 2008-12-04 10:44:18 -0600 (Thu, 04 Dec 2008) | 15 lines
Fix a callerid parsing issue. If someone formatted callerid like the
following: "name <number>" (including the quotation marks), then the parts
would be parsed as
name: "name
number: number
This is because the closing quotation mark was not discovered since the number
and everything after was parsed out of the string earlier. Now, there is a check
to see if the closing quote occurs after the number, so that we can know if we
should strip off the opening quote on the name.
Closes AST-158
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r160938 | mvanbaak | 2008-12-04 17:37:13 +0100 (Thu, 04 Dec 2008) | 9 lines
Add debug flag so skinny debug will show information about packets.
We dont want to scare users with this, so we added a devmode compile flag
(closes issue #13952)
Reported by: wedhorn
Patches:
packetdebug3.diff uploaded by wedhorn (license 30)
Tested by: mvanbaak, wedhorn
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r160760 | murf | 2008-12-03 14:09:15 -0700 (Wed, 03 Dec 2008) | 23 lines
Merged revisions 160703 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r160703 | murf | 2008-12-03 13:41:42 -0700 (Wed, 03 Dec 2008) | 11 lines
(closes issue #13597)
Reported by: john8675309
Patches:
patch.13597 uploaded by murf (license 17)
Tested by: murf, john8675309
This patch causes the setcid func to update the CDR
clid after setting the channel field.
I also notice that in trunk, the num/number of 1.4 is
left out; I decided to include the option to use
either in trunk, so as not to have 1.4 upgraders
not to have problems.
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r160663 | eliel | 2008-12-03 17:25:30 -0200 (Wed, 03 Dec 2008) | 13 lines
- iax2-provision was not freeing iax_templates structure when unloading the chan_iax2.so module.
- Move the code to start using the LIST macros.
Review: http://reviewboard.digium.com/r/72
(closes issue #13232)
Reported by: eliel
Patches:
iax2-provision.patch.txt uploaded by eliel (license 64)
(with minor changes pointed by Mark Michelson on review board)
Tested by: eliel
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r160626 | mmichelson | 2008-12-03 12:37:46 -0600 (Wed, 03 Dec 2008) | 16 lines
Add some safety measures when using gosub, especially when using the options
for app_dial and app_queue to run a gosub when the call is answered.
* Check for the existence of the gosub target in gosub_exec. If it is nonexistent,
then this will cause errors when we attempt to actually run the gosub, including
a definite memory leak and potential crashes. Return an error in this situation
* Check the return value of pbx_exec in app_dial and app_queue before attempting
to actually run the gosub routine. If there was an error, we should not attempt
to run the gosub.
* Change a '|' to a ',' in app_queue.
* Add some extra curly braces where they had been missing previously.
(closes issue #13548)
Reported by: fiddur
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r160585 | tilghman | 2008-12-03 11:59:36 -0600 (Wed, 03 Dec 2008) | 11 lines
Blocked revisions 160570 via svnmerge
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r160570 | tilghman | 2008-12-03 11:55:12 -0600 (Wed, 03 Dec 2008) | 5 lines
During bridge code, the channel bridge may return a retry code, if a transfer
was initiated but not yet completed. If the bridge is immediately retried,
then we may send a storm of TXREQ packets, even though the first set is sent
reliably (retransmitted). Fixes AST-137.
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r160559 | tilghman | 2008-12-03 11:38:59 -0600 (Wed, 03 Dec 2008) | 14 lines
Merged revisions 160558 via svnmerge from
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r160558 | tilghman | 2008-12-03 11:34:34 -0600 (Wed, 03 Dec 2008) | 7 lines
If an entry is added to the directory during a scan when another entry expires,
then that new entry will not be processed promptly, but must wait for either a
future entry to start or a current entry's retry to occur. If no other entries
exist in the directory (other than the new entries) when a bunch expire, then
the new entries must wait until another new entry is added to be processed.
This was a rather weird race condition, really. Fixes AST-147.
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r160555 | mmichelson | 2008-12-03 11:07:09 -0600 (Wed, 03 Dec 2008) | 11 lines
When investigating issue #13548, I found that gosub
handling in app_queue was just completely wrong, mostly
because the channel operations being performed were being
done on the incorrect channel.
With this set of changes, a gosub will correctly run on
the answering queue member's channel. There are still crash
issues which occur if there are dialplan syntax errors, so
I cannot yet close the referenced issue.
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