https://origsvn.digium.com/svn/asterisk/branches/1.4
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r86471 | file | 2007-10-19 12:33:49 -0300 (Fri, 19 Oct 2007) | 6 lines
Fix two issues with domains and transfers. If a port was given in the hostname it was treated as part of the hostname. If domains were configured but external domains were not enabled all transfers would be considered remote.
(closes issue #11027)
Reported by: ramonpeek
Patches:
11027-1.diff uploaded by ramonpeek (license 266)
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r86149 | russell | 2007-10-17 12:57:45 -0500 (Wed, 17 Oct 2007) | 8 lines
If Asterisk is in the middle of shutting down, respond to OPTIONS
with 503 Unavailable.
(closes issue #10994)
Reported by: eserra
Patches:
sip-options-503.patch uploaded by eserra (license 45)
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Don't improperly memset() over an ast_str. This was leftover from before it
got changed to use ast_str.
(closes issue #11003, reported by pj)
(closes issue #10770, reported by yehavi)
(patched by me)
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r85604 | russell | 2007-10-15 11:54:57 -0500 (Mon, 15 Oct 2007) | 6 lines
Make the default for the srvlookup option to be yes. It doesn't really make
sense for it to default to off. The default configuration file has it on, and
proper RFC behavior, as indicated by a comment in the code, is for it to be on.
So, let's have it on by default to make lives easier.
(closes issue #10954, suggested by jtodd)
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r84370 | russell | 2007-10-02 09:12:35 -0500 (Tue, 02 Oct 2007) | 6 lines
Use snprintf instead of sprintf in one place. There is no vulnerability here
due to various buffer sizes around the code, but I still didn't like seeing a
non length-limited copy of data coming off of the wire into a stack buffer, as
this would be a problem in the future if buffer sizes elsewhere got changed or
size limitations removed ...
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a patch for it. It replaces a bunch of simple calls to snprintf with ast_copy_string
(closes issue #10843)
Reported by: Corydon76
Patches:
2007092900_10843.diff uploaded by mvanbaak (license 7)
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r83941 | russell | 2007-09-26 16:15:15 -0500 (Wed, 26 Sep 2007) | 5 lines
Add a log message that was requested by the masses in the developer tutorial
session at Astricon. chan_sip did not output any message when a call was
rejected because the extension was not found. This adds a verbose message
(at verbose level 3) to note when this happens.
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r83432 | russell | 2007-09-21 09:37:20 -0500 (Fri, 21 Sep 2007) | 4 lines
gcc 4.2 has a new set of warnings dealing with cosnt pointers. This set of
changes gets all of Asterisk (minus chan_alsa for now) to compile with gcc 4.2.
(closes issue #10774, patch from qwell)
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Reported by: junky
Patches:
register_trying.diff.txt uploaded by jcmoore
Disable sending 100 Trying on REGISTER attempts and make it an option. This has been signed off by oej.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82257 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: casper
Patches:
chan_sip.c.82076.diff uploaded by casper (license 55)
Remove double check for zombie flag and optimize things a bit.
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r81832 | russell | 2007-09-06 17:28:57 -0500 (Thu, 06 Sep 2007) | 16 lines
(closes issue #9724, closes issue #10374)
Reported by: kenw
Patches:
9724.txt uploaded by russell (license 2)
Tested by: kenw, russell
Resolve a deadlock that occurs when doing a SIP transfer to parking.
I come across this type of deadlock fairly often it seems. It is very important
to mind the boundary between the channel driver and the core in respect to the
channel lock and the channel-pvt lock. Channel drivers lock to lock the
pvt and then the channel once it calls into the core, while the core will do
it in the opposite order. The way this is avoided is by having channel drivers
either release their pvt lock while calling into the core, or such as in this
case, unlocking the pvt just long enough to acquire the channel lock.
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r81442 | kpfleming | 2007-09-04 11:40:39 -0500 (Tue, 04 Sep 2007) | 2 lines
there is no point in sending 401 Unauthorized to a UAS that sent us a properly-formatted Authentication header with the expected username and nonce but an incorrect response (which indicates the shared secret does not match)... instead, let's send 403 Forbidden so that the UAS doesn't retry with the same authentication credentials repeatedly
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r81395 | file | 2007-08-30 18:23:50 -0300 (Thu, 30 Aug 2007) | 6 lines
(closes issue #10514)
Reported by: casper
Patches:
chan_sip.c.80129.diff uploaded by casper (license 55)
Remove needless check for AUTH_UNKNOWN_DOMAIN. It was impossible for it to ever be that value.
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Reported by: tootai
Make sure the external IP address has the standard SIP port set for when the user does not specify the port in the externip setting.
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r81012 | file | 2007-08-27 12:01:59 -0300 (Mon, 27 Aug 2007) | 6 lines
(closes issue #10561)
Reported by: jesselang
Patches:
chan_sip-ChannelReload-20080825.patch uploaded by jesselang (license 202)
Remove an extra \r\n to make the ChannelReload event conform with every other event.
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r80183 | russell | 2007-08-21 13:42:15 -0500 (Tue, 21 Aug 2007) | 7 lines
Don't record SIP dialog history if it's not turned on. Also, put an upper
limit on how many history entires will be stored for each SIP dialog. It is
currently set to 50, but can be increased if deemed necessary.
(closes issue #10421, closes issue #10418, patches suggested by jmoldenhauer,
patches updated by me)
(Security implications documented in AST-2007-020)
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(closes issue #10430)
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r79904 | qwell | 2007-08-17 14:12:19 -0500 (Fri, 17 Aug 2007) | 11 lines
Don't send a semicolon over the wire in sip notify messages.
Caused by fix for issue 9938.
I basically took the code that existed before 9938 was fixed, and
copied it into a new function - ast_unescape_semicolon
There should be very few places this will be needed (pbx_config
does NOT need this (see issue 9938 for details))
Issue 10430, patch by me, with help/ideas from murf (thanks murf).
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in place of a very common construct. I also used it in a number of places
in chan_sip.
if (id > -1)
ast_sched_del(sched, id);
id = ast_sched_add(sched, ...);
changes to:
ast_sched_replace(id, sched, ...);
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r79857 | russell | 2007-08-17 08:37:08 -0500 (Fri, 17 Aug 2007) | 5 lines
Fix some crashes in chan_sip. This patch changes various places that add items
to the scheduler to ensure that they don't overwrite the ID of a previously
scheduled item. If there is one, it should be removed.
(closes issue #10391, closes issue #10256, probably others, patch by me)
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r79523 | file | 2007-08-15 11:18:44 -0300 (Wed, 15 Aug 2007) | 6 lines
(closes issue #10456)
Reported by: irroot
Patches:
sip_timeout.patch uploaded by irroot (license 52)
Change hardcoded timer value to defined value. I'm doing this in 1.4 as well so if it needs to be changed in the future this place would not have been forgotten.
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r79174 | file | 2007-08-13 11:18:04 -0300 (Mon, 13 Aug 2007) | 4 lines
(closes issue #10437)
Reported by: haklin
Don't set the callerid name and number a second time on a newly created channel. ast_channel_alloc itself already sets it and setting it twice would cause a memory leak.
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