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r77794 | russell | 2007-07-30 15:16:43 -0500 (Mon, 30 Jul 2007) | 8 lines

Fix an issue that could potentially cause corruption of the global iax frame
queue.  In the network_thread() loop, it traverses the list using the
AST_LIST_TRAVERSE_SAFE macro.  However, to remove an element of the list within
this loop, it used AST_LIST_REMOVE, instead of AST_LIST_REMOVE_CURRENT, which I
believe could leave some of the internal variables of the SAFE macro invalid.
Mihai says that he already made this change in his local copy and it didn't help
his VNAK storm issues, but I still think it's wrong.  :)

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(closes issue #10083)
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r77795 | qwell | 2007-07-30 15:17:08 -0500 (Mon, 30 Jul 2007) | 6 lines

Applications like SayAlpha() should not hang up the channel if you
 request an "unknown" character such as a comma.
Instead, skip the character and move on.

Issue 10083, initial patch by jsmith, modified by me.

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Being bitmasks, it is a lot easier to read this way.



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 * Use a thread local ast_str for building the string that will be written out
   to the console for debug, and to the FD for the AGI itself, instead of allocating
   a buffer on the heap every time the function is called.
 * Use the information contained within the ast_str to determine how many bytes
   need to be written instead of calling strlen().


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inside of a function.  (Yes, it would!)  Replace it with a note that explains
why it can't be done using the way that the AST_THREADSTORAGE macro is
currently defined.


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r77788 | russell | 2007-07-30 14:13:31 -0500 (Mon, 30 Jul 2007) | 10 lines

(closes issue #10279)
Reported by: seanbright
Patches:
      res_agi.carefulwrite.1.4.07252007.patch uploaded by seanbright (license 71)
      res_agi.carefulwrite.trunk.07252007.patch uploaded by seanbright (license 71)

Allow the "agi_network: yes" line to be printed out in the AGI debug output.
Also, allow partial writes to be handled when writing out this line just like
it is for all of the others.

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r77785 | russell | 2007-07-30 13:55:15 -0500 (Mon, 30 Jul 2007) | 3 lines

file and I both committed changes for issue #10301.  Remove a duplicated
assignment to restore the original value of the previous channel.

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r77783 | tilghman | 2007-07-30 13:43:55 -0500 (Mon, 30 Jul 2007) | 10 lines

Merged revisions 77782 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r77782 | tilghman | 2007-07-30 13:40:54 -0500 (Mon, 30 Jul 2007) | 2 lines

Revert change in revision 71656, even though it fixed a bug, because many people were depending upon the (broken) behavior.

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r77780 | russell | 2007-07-30 12:29:43 -0500 (Mon, 30 Jul 2007) | 16 lines

(closes issue #10301)
Reported by: fnordian
Patches:
      asterisk-1.4.9-channel.c.patch uploaded by fnordian (license 110)
      Additional changes by me

Fix some problems in channel_find_locked() which can cause an infinite loop.
The reference to the previous channel is set to NULL in some cases.  These changes
ensure that the reference to the previous channel gets restored before needing
it again.

I'm not convinced that the code that is setting it to NULL is really the right
thing to do.  However, I am making these changes to fix the obvious problem
and just leaving an XXX comment that it needs a better explanation that what
is there now.

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r77778 | file | 2007-07-30 14:11:02 -0300 (Mon, 30 Jul 2007) | 4 lines

(closes issue #10327)
Reported by: kkiely
Instead of directly mucking with the extension/context/priority of the channel we are transferring when it has a PBX simply call ast_async_goto on it. This will ensure that the channel gets handled properly and sent to the right place.

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r77771 | file | 2007-07-30 12:47:52 -0300 (Mon, 30 Jul 2007) | 6 lines

(closes issue #10301)
Reported by: fnordian
Patches:
      asterisk-1.4.9-channel.c.patch uploaded by fnordian (license 110)
Restore previous behavior where if we failed to lock the channel we wanted we would return to exactly the same point as if we had just reentered the function.

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r77768 | file | 2007-07-30 11:51:44 -0300 (Mon, 30 Jul 2007) | 12 lines

Merged revisions 77767 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r77767 | file | 2007-07-30 11:50:02 -0300 (Mon, 30 Jul 2007) | 4 lines

(closes issue #10334)
Reported by: ramonpeek
Pass through the return value from macro_exec through the MacroIf application.

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+ place the link field at the beginning of struct sip_pvt, 
  and not somewhere in the middle;
 
+ in __sip_reliable_xmit, remove a duplicate assignment, and
  put the statements in a more logical order (i.e. first copy
  the payload and associated info, then copy arguments from the
  caller, then finish initializing the headers...)
 
nothing to backport.



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was misleading - the function deals with all incoming packets, be
them requests or responses.



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SIP_NOVIDEO, SIP_DIALOG_ANSWEREDELSEWHERE, SIP_PAGE2_NOTEXT,
SIP_PAGE2_OUTGOING_CALL
These are seldom used so the diff is relatively small.

Note that 'OUTGOING_CALL' is dangerously similar to another
dialog flag, 'SIP_OUTGOING', so the description will need to
clarify the different meaning of the two.

Also note that the description of NOTEXT is a bit unclear - does
it mean we don't support it, or 'not requested or not supported' ?


On passing fix a comment referring to video instead of text.

Finally, mark with XXX a possibly misleading debugging message.
(maybe the latter is worth backporting).



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CLI lines. This helps maintaining consistency on output, slightly
improves readability, and maybe one day will make it easier to
translate the output in other languages (though i have a hard time
believing that a CLI user who needs 'yes' and 'no' to be translated
can actually figure out what he/she is doing!)



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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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Remove definitions for now-unused flags, and add references to print
routines for other flags.



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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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