Start putting these variables in a single struct (called 'sip_cfg' for the time
being, but it could as well be 'global' or some other name) so it
is easy, when reading the code, to figure out what they are for.
The downside of using struct fields instead of individual global
variables is that the compiler cannot tell if there are unused fields.
But the advantage of not cluttering the namespace and manilpulating
all these variables at once certainly overcome the disadvantagess.
Nothing to backport, again.
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at load time instead of duplicating the initializer.
This should remove the risk of forgetting fields in the
initializer.
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use AST_FORMAT_AUDIO_MASK instead of playing with AST_FORMAT_MAX_AUDIO
to determine audio formats.
There is a dubious use of AST_FORMAT_MAX_AUDIO in sip_request_call()
which surely needs fixing, namely:
/* mask request with some set of allowed formats.
* XXX this needs to be fixed.
* The original code uses AST_FORMAT_AUDIO_MASK, but it is
* unclear what to use here. We have global_capabilities, which is
* configured from sip.conf, and sip_tech.capabilities, which is
* hardwired to all audio formats.
*/
The latter is possibly something to backport when fixed.
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Move together flags used in the same way (e.g. dialog only,
dialog-peer, ...) so it will become easier to deal with them
in a more systematic way.
This is being done in stages so it will be easier to detect
breakage, if any should occur.
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the original pointer while calling the function.
on passing add some comments on one of the places where it
is used, and explain why it is safe there.
again, a no-op for practical purposes.
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dialog_ref/unref (they are a no-op at the moment).
Also clean a pointer after freeing memory to avoid
dangling references, and write a for() loop in canonical form.
In practice, everything in this commit is a no-op.
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This commit is, for all practical purposes, a no-op, as it only
introduces the dialog_ref() and dialog_unref() methods, and uses them
in a few places (not all the places where they would be needed).
The goal is to start annotating the code with these calls, so the transition
to a proper container will be easier.
Nothing to backport.
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In a nutshell, these fields are used to tell a sip entity
the address and port its request came from, and are extremely
useful in the presence of NATs, especially with symmetric NATs
where STUN is totally ineffective.
This patch stores the address and port in the 'ourip' field of
the dialog descriptor, so they can be reused in subsequent transactions.
As it is, it works well for things like REGISTER requiring authentication,
because the second REGISTER request (with auth credentials) will carry
the correct address. Maybe it can also be useful, in case of an address
change, to do one or both of the following:
+ propagate the new address to the parent user/peer descriptor so that new
dialogs will use the correct address from the beginning.
This is trivial to implement, I am just waiting for feedback on this.
+ re-issue a request in case of an address change. This a lot less trivial,
maybe unnecessary, and probably covered by the previous item.
I would seriously consider this patch for addition to 1.4 and 1.2.
The code is very little intrusive, and it would solve in a correct
way the nat traversal problems for which externip/externaddr/stunaddr
are only a partial and expensive workaround.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r77536 | file | 2007-07-27 13:27:16 -0300 (Fri, 27 Jul 2007) | 6 lines
(closes issue #10323)
Reported by: julianjm
Patches:
chan_sip_device_state_hold_fix.v1.diff.txt uploaded by julianjm (license 99)
Clear ONHOLD flag when decrementing the onHold peer count. If we did not do this the count may keep decreasing.
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r77460 | file | 2007-07-26 20:19:04 -0300 (Thu, 26 Jul 2007) | 4 lines
(closes issue #10302)
Reported by: litnialex
If a DTMF end frame comes from a channel without a begin and it is going to a technology that only accepts end frames (aka INFO) then use the minimum DTMF duration if one is not in the frame already.
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r77410 | russell | 2007-07-26 16:23:23 -0500 (Thu, 26 Jul 2007) | 10 lines
AST_DEVMODE was defined in trunk, but not in 1.4. When Asterisk is compiled
under dev mode, AST_DEVMODE will get defined in buildopts.h. Change 1.4 to
define it in the same way that trunk does. Also, revert the change that added
this define in the Makefile
The advantage to doing it this way is that buildopts.h gets installed when
you install Asterisk. Then, when building any out of tree modules, or
building asterisk-addons, these modules know which options the rest of Asterisk
was built with.
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r77380 | mmichelson | 2007-07-26 15:35:17 -0500 (Thu, 26 Jul 2007) | 7 lines
Fixes to get ast_backtrace working properly. The AST_DEVMODE macro was never defined so the majority of ast_backtrace never
attempted compilation. The makefile now defines AST_DEVMODE if configure was run with --enable-dev-mode. Also, changes were
made to acccomodate 64 bit systems in ast_backtrace.
Thanks to qwell, kpfleming, and Corydon76 for their roles in allowing me to get this committed
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r77318 | mmichelson | 2007-07-26 13:30:29 -0500 (Thu, 26 Jul 2007) | 8 lines
Two consecutive calls to PQfinish could occur, meaning free gets called on the same variable twice.
This patch sets the connection to NULL after calls to PQfinish so that the problem does not occur.
Also in this patch, prashant_jois informed me that it is safe to pass a null pointer to PQfinish, so
I have removed the check for conn's existence from my_unload_module.
(closes issue 10295, reported by junky, patched by me with input from prashant_jois)
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r77191 | murf | 2007-07-25 16:39:27 -0600 (Wed, 25 Jul 2007) | 1 line
This fix solves problem with intense squelch noise when someone joins conf in bug 9430; We repro'd the problem with meetme opts of 'CciMo'; Josh Colp supplied this patch, and I'm applying it. It looks like playing the recorded username will louse up the next thing played into the channel. Josh rearranged the code so as to start things over before playing data directly into the conference.
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r77176 | file | 2007-07-25 19:16:10 -0300 (Wed, 25 Jul 2007) | 4 lines
(closes issue #10303)
Reported by: jtodd
Add SPEECH_DTMF_TERMINATOR variable so the user can specify the digit to terminate a DTMF string with. If none is specified then no terminator will be used.
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At first sight (but the function is very large so i am not 100% sure)
the code seems correct, so maybe my compiler is just not smart
enough to figure that out at the optimization level it has.
Not worthwhile merging to 1.4 i believe.
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Luigi Rizzo
Tilghman Lesher
Joshua Colp
Steve Murphy
Mark Michelson
Kevin P. Fleming
Russell Bryant