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f38db490-d61c-443f-a65b-d21fe96a405b
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blitzrage's test machines. It was one of the situations where he was seeing
hung channels, and may be the cause of some of the reports from other people.
(related to issue #9235)
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SpeechBackground.
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incoming call on the trunk, or if the trunk reached its ring timeout.
This patch changes the variable to say "RINGTIMEOUT" in that case.
(issue #9973, reported by n00dle, patch by me)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r69469 | qwell | 2007-06-14 18:21:45 -0500 (Thu, 14 Jun 2007) | 4 lines
Fix an issue where the line number in an unterminated comment block error message would show the wrong line number.
"Reported" to me on #asterisk (somebody posted an error message, and I happened to catch it)
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(issue #9964, ljmid)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r69258 | qwell | 2007-06-14 10:15:53 -0500 (Thu, 14 Jun 2007) | 4 lines
Change a quite broken while loop to a for loop, so "continue;" works as expected instead of eating 99% CPU...
Issue 9966, patch by me.
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reported by simone cittadini)
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outside of the block that checks for FLAG_FATAL to be set. This flag is only
set when the packet is transmitted with the reliability set to XMIT_CRITICAL
when the original packet is transmitted. A BYE is always sent with it set
to XMIT_RELIABLE, meaning this code could never be encountered. This resulted
in seeing some SIP channels that would never go away with the last packet
sent being a BYE.
(part of issue #9235, patch from jcmoore)
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voicemail emails and attachments.
This fix has also been tested on Thunderbird, Evolution, Pine, and Mutt.
(Issue 9336, reported by marwick, patched by mutterc)
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(issue #9912 reported by junky)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r69127 | file | 2007-06-13 14:12:48 -0400 (Wed, 13 Jun 2007) | 2 lines
Return group counting to previous behavior where you could only have one group per category. (issue #9711 reported by irroot)
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helpful when following the logic to debug problems like 9235.
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This issue was reported to me via email by Dmitry Mishchenko. Thanks!
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not called on the result structure after doing a PQexec(). Also, fix up some
formatting in passing.
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it does not collide with the LAGRQ.
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formats already match up. There are code paths that call this function on a
pair of channels multiple times. This made calls fail that were using g729
in some cases. The reason is that codec_g729a will unregister itself from the
list of available translators will all licenses are in use. So, the first
time the function got called, the right translation path was allocated.
However, the second time it got called, the code would not find a translation
path to/from g729 and make the call fail, even if the channel actually already
had a g729 translation path allocated.
(SPD-32)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r68921 | file | 2007-06-12 10:18:57 -0400 (Tue, 12 Jun 2007) | 2 lines
Bring RTP back to Asterisk at the end of a native bridge no matter what.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r68732 | crichter | 2007-06-11 18:49:00 +0200 (Mo, 11 Jun 2007) | 1 line
added check for NULL Pointer when calling misdn_new. Asterisk does not allow us to create channels anymore when stop gracefully is used :). also modified the restart_indicator to 0
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r68682 | file | 2007-06-11 10:29:58 -0400 (Mon, 11 Jun 2007) | 2 lines
Improve deadlock handling of the channel list. (issue #8376 reported by one47)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r68631 | crichter | 2007-06-11 11:18:01 +0200 (Mo, 11 Jun 2007) | 1 line
fixed problem that the dummybc chanels had no lock, checking for the lock now. Also fixed the channel restart stuff, we can now specify and restart particular channels too.
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r68526 | russell | 2007-06-08 17:22:36 -0500 (Fri, 08 Jun 2007) | 4 lines
Don't automatically hang up after running Dictate so that callers can exit
cleanly using '#'
(closes issue #9577, patch from Thomas Andrews)
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r68397 | file | 2007-06-07 20:15:33 -0400 (Thu, 07 Jun 2007) | 2 lines
Don't call ast_waitstream_full when the control file descriptor and audio file descriptor are not set, simply call ast_waitstream! (issue #8530 reported by rickead2000)
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r68368 | file | 2007-06-07 19:59:04 -0400 (Thu, 07 Jun 2007) | 2 lines
Do a DNS lookup immediately upon calling the dnsmgr function, don't wait until a refresh happens. (issue #9097 reported by plack)
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r68351 | russell | 2007-06-07 18:13:33 -0500 (Thu, 07 Jun 2007) | 3 lines
Fix a problem where saying a character wouldn't properly break out when the caller pressed '#'
(issue #8113, reported by patbaker82, patch from jamesgolovich (hey, long time no see!) and patbaker82)
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Request for feedback was sent to asterisk-dev mailing list, with little response.
Issue 9118, patch by junky.
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- use inaddrcmp(), since we have it
- output the type of frame and subclass being dropped, and the type/subclass that is already being processed (which caused the drop)
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queues.
Also, remove an unneeded leading slash for the astdb family.
(issue #9911, patch by atis)
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correct length.
Issue 9887, patch by DEA.
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r68204 | qwell | 2007-06-07 15:02:50 -0500 (Thu, 07 Jun 2007) | 4 lines
Don't try to save voicemail greetings unless the user presses '1' to accept/save.
Issue 9904, patch by me.
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get stored properly.
(Issue 8016, reported by edhorton, patched by alamantia with modification by me. Thanks to Jason Parker
for the advice on this).
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(issue #9381)
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want to start from a specific point in the channel list. (issue #9324 reported by slavon)
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r68070 | file | 2007-06-07 10:19:40 -0400 (Thu, 07 Jun 2007) | 2 lines
Allow the 'g' option to work if used with the 'S' option. (issue #9888 reported by gasparz)
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using TLS, but
if you have initialized TLS at all during the lifetime of the module. So if you reload
to disable TLS, it won't help.
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prevent
crashes while we are trying to find a workaround.
Iksemel development seems to have stalled and we might have to stop using the
TCP/TLS connections in that library and use our own, which would scale better
from a poll/select perspective I guess. It would also make it easier to migrate
to OpenSSL and stop Asterisk from depending on both OpenSSL and GnuTLS.
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