Copy the outbound proxy IP address into the SIP dialog structure as the IP address we will
be sending to. This has to be done because the logic that determines what local IP address to use
in the SIP messages is not aware of an outbound proxy being in place. It only knows what IP address
we are sending to.
(closes issue #12006)
Reported by: mnicholson
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@188067 f38db490-d61c-443f-a65b-d21fe96a405b
This allows for the variables to be accessed if a member macro is run.
Thanks to Grigoriy Puzankin for bringing this up on the -dev list.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@188032 f38db490-d61c-443f-a65b-d21fe96a405b
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r187962 | jpeeler | 2009-04-10 17:16:13 -0500 (Fri, 10 Apr 2009) | 9 lines
Fix module embedding for chan_h323.
Include libchanh323.a in the modules.link file so that all the symbols can be
resolved at link time.
(closes issue #11966)
Reported by: dome
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187963 f38db490-d61c-443f-a65b-d21fe96a405b
Include libchanh323.a in the modules.link file so that all the symbols can be
resolved at link time.
(closes issue #11966)
Reported by: dome
Patches:
issue_11966.patch uploaded by kpfleming (license 421)
Tested by: jpeeler
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187906 f38db490-d61c-443f-a65b-d21fe96a405b
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r187865 | russell | 2009-04-10 14:26:40 -0500 (Fri, 10 Apr 2009) | 4 lines
Support "signaling" in addition to "signalling".
The sample configuration file has references to both spellings.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187866 f38db490-d61c-443f-a65b-d21fe96a405b
The code will now only change the address and port. It will not overwrite any other values.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187773 f38db490-d61c-443f-a65b-d21fe96a405b
This allows for you to change the From header for outgoing MWI
NOTIFY requests. Prior to this, the best you could do was to
set a callerid in the general section of sip.conf. The problem
was that this was used for all outbound requests, not just
MWI NOTIFY requests.
AST-201
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187560 f38db490-d61c-443f-a65b-d21fe96a405b
The F option to app_dial has been modified to accept no parameters and perform
the above functionality. I don't see anywhere else that is doing function
overloading, but this really is the best place for this operation because:
- It makes it close to the 'g' option in the argument list which provides
similar functionality.
- The existing code to support the current F option provides a very
convienient location to add this new feature.
(closes issue #12381)
Reported by: michael-fig
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187491 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r187484 | mmichelson | 2009-04-09 13:51:20 -0500 (Thu, 09 Apr 2009) | 18 lines
Handle a SIP race condition (reinvite before an ACK) properly.
RFC 5047 explains the proper course of action to take if a
reINVITE is received before the ACK from a previous invite
transaction. What we are to do is to treat the reINVITE as
if it were both an ACK and a reINVITE and process it normally.
Later, when we receive the ACK we had been expecting, we will
ignore it since its CSeq is less than the current iseqno of
the sip_pvt representing this dialog.
(closes issue #13849)
Reported by: klaus3000
Patches:
13849_v2.patch uploaded by mmichelson (license 60)
Tested by: mmichelson, klaus3000
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187488 f38db490-d61c-443f-a65b-d21fe96a405b
get_cid_name should not be called with a channel lock. get_cid_name calls ast_get_hint which eventually calls pbx_find_extension. pbx_find_extension starts and stops autoservice which should not be done with a channel lock, so get_cid_name should not be called with one.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187426 f38db490-d61c-443f-a65b-d21fe96a405b
The moh_register function links an mohclass and then immediately
unrefs the class since the container now has a reference. The problem
with using realtime music on hold is that the class is allocated,
registered, and started in one fell swoop. The refcounting logic
resulted in the count being off by one. The same problem did not
happen when using a static config because the allocation and registration
of an mohclass is a separate operation from starting moh. This also did
not affect non-cached realtime moh because the classes are not registered
at all.
I also have modified res_musiconhold to use the _t_ variants of the ao2_
functions so that more info can be gleaned when attempting to trace the
refcounts. I found this to be incredibly helpful for debugging this issue
and there's no good reason to remove it.
(closes issue #14661)
Reported by: sum
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187421 f38db490-d61c-443f-a65b-d21fe96a405b
This was accomplished using a set of options and the setoption channel callback.
The core calls into the channel driver using these options and the channel driver
either returns success or failure.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187360 f38db490-d61c-443f-a65b-d21fe96a405b
(inspired by this evening's Toronto Asterisk Users Group meeting and previous dicussions amongst various community members)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187269 f38db490-d61c-443f-a65b-d21fe96a405b
The biggest change done here was elimination of the backup_config for use with
features. Previously, the bridging code upon detecting a feature would set the
start time of the bridge to the start time of the feature. Then after the
feature had either expired or timed out the start time would be reset to the
true bridge start time from the backup_config. Now, the time differences are
calculated with respect to the newly added feature_start_time timeval instead.
There should be no behavior changes from the previous functionality aside from
the bridge timing being unaffected by either valid or partial feature matches.
Previously the timing would be increased by the length of time configured for
featuredigittimeout, which was probably never noticed.
(closes issue #14503)
Reported by: KNK
Tested by: jpeeler
Review: http://reviewboard.digium.com/r/179/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187211 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r187209 | tilghman | 2009-04-08 15:39:13 -0500 (Wed, 08 Apr 2009) | 4 lines
Backport resolution for file descriptor leak in 1.6.0 to 1.4.
This fixes short reads in http manager sessions, such as those done by the
ast-gui branch. (Fixes AST-198)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187210 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r187045 | mmichelson | 2009-04-08 11:52:03 -0500 (Wed, 08 Apr 2009) | 10 lines
Fix a small logical error when loading moh classes.
We were unconditionally incrementing the number of mohclasses
registered. However, we should actually only increment if the
call to moh_register was successful.
While this probably has never caused problems, I noticed it
and decided to fix it anyway.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187046 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r186984 | mmichelson | 2009-04-08 10:26:46 -0500 (Wed, 08 Apr 2009) | 24 lines
Make a couple of changes with regards to a new message printed in ast_read().
"ast_read() called with no recorded file descriptor" is a new message added
after a bug was discovered. Unfortunately, it seems there are a bunch of places
that potentially make such calls to ast_read() and trigger this error message
to be displayed. This commit does two things to help to make this message appear
less.
First, the message has been downgraded to a debug level message if dev mode is
not enabled. The message means a lot more to developers than it does to end users,
and so developers should take an effort to be sure to call ast_read only when
a channel is ready to be read from. However, since this doesn't actually cause an
error in operation and is not something a user can easily fix, we should not spam
their console with these messages.
Second, the message has been moved to after the check for any pending masquerades.
ast_read() being called with no recorded file descriptor should not interfere with
a masquerade taking place.
This could be seen as a simple way of resolving issue #14723. However, I still want
to try to clear out the existing ways of triggering this message, since I feel that
would be a better resolution for the issue.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@186985 f38db490-d61c-443f-a65b-d21fe96a405b
doxyref.h was created to hold miscellaneous documentation that was not specific
to a part of the code. This file has grown quite a bit so I decided to start
splitting parts of it out into new files. Now, you can drop a new file into
include/asterisk/doxygen/ and it will be processed by doxygen.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@186953 f38db490-d61c-443f-a65b-d21fe96a405b
While browsing chan_sip the other day, I noticed this dangerous code in
dialog_needdestroy(). This function is an ao2_callback. It is absolutely
_not_ okay to unlock the container from within this function. It's also not
clear why it was useful. Given that it could cause memory corruption, I have
removed it.
There was also a TODO comment left describing a potential implementation of
an improvement to the needdestroy handling. I'm not convinced that what was
described is the best choice here, so I have briefly described the way that
this function is used today that could be improved.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@186928 f38db490-d61c-443f-a65b-d21fe96a405b
tilghman