Steve Murphy [Thu, 26 Jul 2007 01:33:55 +0000 (01:33 +0000)]
The upgrade of application argument separators to comma has an effect on AEL; I commented out the code that substitutes commas with vertbars, so we can get apps to parse their args correctly.
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.
........
(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.
Luigi Rizzo [Wed, 25 Jul 2007 21:58:13 +0000 (21:58 +0000)]
silence a warning in ast-devmode on a potentially uninitialized var.
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.
Added a membercount variable to call_queue struct which keeps track of the number of logged in members in a particular queue.
This makes it so that the 'n' option for Queue() can act properly depending on which strategy is used. If the strategy is
roundrobin, rrmemory, or ringall, we want to ring each phone once before moving on in the dialplan. However, if any other strategy is
used, we will only ring one phone since it cannot be guaranteed that a different phone will ring on subsequent attempts to ring a phone.
As a side effect of this, the QUEUE_MEMBER_COUNT dialplan function now just reads the membercount variable instead of traversing through
the member list to figure out how many members there are.
Special thanks to blitzrage for helping to test this out.
(closes issue #10127, reported by bcnit, patched by me, tested by blitzrage)
Jason Parker [Tue, 24 Jul 2007 15:35:58 +0000 (15:35 +0000)]
The chan_skinny Dial() syntax was funky. You had to do Dial(Skinny/line@device)
This allows you to just Dial(Skinny/line), as long as line isn't ambiguous.
Note that this does not remove or deprecate the "old" syntax, as it's still
quite useful - even moreso if shared lines get implemented.
Initial patch by me, with some changes and suggestions from wedhorn.
(closes issue #10263)
Luigi Rizzo [Tue, 24 Jul 2007 14:49:49 +0000 (14:49 +0000)]
two small fixes when using stun (reported by Marta Carbone):
+ externexpire was not initialized properly;
+ stunaddr was not handled properly on a sip reload
You need to put static in front of a static RWLIST declaration to make it really static... and don't call AST_RWLIST_HEAD_DESTROY on a statically declared list.
It was our stated intention for 1.4 that files created in app_voicemail should
depend upon the umask. Unfortunately, mkstemp() creates files with mode 0600,
regardless of the umask. This corrects that deficiency.
Enhance AGI with several fixes:
- Makes the structures handling external AGI commands a bit more thread-safe
- Makes AGI transparently work with both live and hungup channels
- DeadAGI is hence no longer necessary and is deprecated
- CLI bug fixes
- Commands will refuse to run if the channel is dead and the command is nonsensical
for dead channels.
Merge the dialplan_aesthetics branch. Most of this patch simply converts applications
using old methods of parsing arguments to using the standard macros. However, the big
change is that the really old way of specifying application and arguments separated by
a comma will no longer work (e.g. NoOp,foo|bar). Instead, the way that has been
recommended since long before 1.0 will become the only method available (e.g. NoOp(foo,bar).
(closes issue #5866)
Reported by: tyler
Do not force channel format changes when a generator is present. The generator may have changed the formats itself and changing them back would cause issues.
Luigi Rizzo [Mon, 23 Jul 2007 14:45:22 +0000 (14:45 +0000)]
add two missing entries in the replica of the sip_tech that
does not use DTMF BEGIN frames.
1.4 seems correct (it does not have the two fields).
However, as this bug shows, the current way of creating the sip_tech
replica is too error-prone, one can easily forget to update one of
the two entries. Perhaps it would be better to create sip_tech_info
expliclty at module load, by doing
Russell Bryant [Mon, 23 Jul 2007 14:21:41 +0000 (14:21 +0000)]
(closes issue #10192)
Reported by: bbryant
Patches:
20070720__core_debug_by_file.patch uploaded by bbryant (license 36)
(with some modifications by me)
Tested by: russell, bbryant
This set of changes introduces the ability to set the core debug or verbose
levels on a per-file basis. Interestingly enough, in 1.4, you have the ability
to set core debug for a single file, but that functionality was accidentally
lost in the conversion of the CLI commands to the new format.
This patch improves upon what was in 1.4 by letting you set it for more than 1
file, and by also supporting verbose.
*** Janitor Project ***
This patch also introduces a new macro, ast_verb(), which is similar
to ast_debug(). Setting the per file verbose value only works for messages that
use this macro. Converting existing uses of ast_verbose() can be done like:
if (option_debug > 2)
ast_verbose(VERBOSE_PREFIX_3 "Something useful\n");
Luigi Rizzo [Mon, 23 Jul 2007 14:18:04 +0000 (14:18 +0000)]
introduce two functions, map_x_s() and map_s_x(), to map
between integers and strings using a single translation table,
and use them in a few places instead of ad-hoc routines
that duplicate the table.
On passing, note that REFER_CONFIRMED is never used, and add a
few comments.
(closes issue #10268)
Reported by: mvanbaak
Patches:
chan_skinny_openbsd.diff uploaded by mvanbaak (license 7)
Add another OS that has to use the Macros for byte ordering.
Use a signed integer for storing the number of bytes in the packet read from
the network. Using an unsigned value here made it impossible to handle an
error returned from recvfrom(). Furthermore, in the case that recvfrom()
did return an error, this would cause a crash due to a heap overflow.
(closes issue #10265, reported by and fix suggested by timrobbins)
Luigi Rizzo [Sun, 22 Jul 2007 19:08:37 +0000 (19:08 +0000)]
comment and slightly restructure handle_request() in the part that handles
responses, so that there is a common exit point.
Mark two places where probably we could return -1 instead of 0 to report
an error to the caller.
(change triggered by investigations on how the 'SIP_PKT_IGNORE' field was used).
Luigi Rizzo [Sun, 22 Jul 2007 18:41:57 +0000 (18:41 +0000)]
Cleanup of flags used in struct sip_request, moving them to
individual variables. Apart from SIP_PKT_IGNORE which was used
a zillion times, the other two are used seldom.
On passing:
- move the arrays to the end of struct sip_request, so a (small)
buffer overflow is less likely to overwrite the other fields;
- note that the 'ignore' argument to handle_invite_replaces() is not
used and should be removed (will be done in a separate commit).
Luigi Rizzo [Sun, 22 Jul 2007 08:42:24 +0000 (08:42 +0000)]
document the way sipdebug works, and implement it through
variables and not flags.
NOTE:
The old behaviour (preserved in this commit) is that if sipdebug
is set in the config file, it can only be disabled by reloading the
config. I am not sure if this is accidental or voluntary, but it
is really unconvenient and I think it should be handled in the same
way as other options i.e. consider requests from the config file
or the cli (or the command line) to be fully equivalent and act on
the same status variable.
Luigi Rizzo [Sat, 21 Jul 2007 17:32:00 +0000 (17:32 +0000)]
Add a note to document how the temporary 'pvt' should be initialized
before using it.
I am unclear on the details right now so i hope someone can comment
more. The obvious (and lazy) approach would be to bzero() all of it
(except for the string pool), but isn't that too much work ?
Feedback wanted here...
Luigi Rizzo [Sat, 21 Jul 2007 03:53:17 +0000 (03:53 +0000)]
use explicit variables for things that don't need to
be stored in ast_flags. First victim is 'SIP_NO_HISTORY'
replaced by a 'do_history' field in the sip_pvt structure.
Luigi Rizzo [Sat, 21 Jul 2007 02:33:25 +0000 (02:33 +0000)]
Use ast_str_append() instead of ast_build_string() to construct
the sdp messages. Overall the code is slightly more readable
(because the string is fully described by a single pointer),
and more efficient (because the length is stored explicitly
so you don't need to do strlen()).
(I have been using this code for almost a year now.)
I wish we had infix string operators to do this sort of things!
Backport a fix for a memory leak that was fixed in trunk in reivision 76221
by rizzo. The memory used for the localaddr list was not freed during a
configuration reload.
Luigi Rizzo [Sat, 21 Jul 2007 01:25:13 +0000 (01:25 +0000)]
We have two 'technology' descriptors for a SIP channel, so
define and use a macro to determine whether we are pointing to
one of them, so when one goes away (or a new one appears) we don't
have to touch all the code.
Luigi Rizzo [Sat, 21 Jul 2007 01:01:10 +0000 (01:01 +0000)]
Enhance NAT support as discussed on the -dev list, i.e.:
+ extensive documentation changes both in sip.conf.sample and in the source;
+ allow "externip" and "externhost" to include a port number as well;
+ allow "bindaddr" to have a port number (making bindport unnecessary,
even though it is still present for backward compatibility);
+ introduce the new "stunaddr" parameter to specify an STUN server to
be used from the main SIP socket;
+ extend the "sip show settings" output to show all the above.
Internally:
+ change related data structures from struct in_addr to struct sockaddr_in
to store the port numbers as well;
+ reorganize ast_sip_ouraddrfor() (should also be renamed to sip_ouraddrfor()
because it is not a generic API, though it might become so if called with
a socket as an additional argument, in which case it can be moved elsewhere).
As mentioned in the documentation, media sessions still do not use STUN so the
port numbers may still be incorrect when Asterisk is behind a NAT
On passing, some of the debugging messages printing media addresses are
probably using the wrong values, but this will be checked/fixed in a
subsequent commit if needed.
Part of the following chunk in the function that handles a "sip reload" is
probably needed on previous versions as well, to avoid leaking the memory
used for the "localaddr" list:
Jason Parker [Fri, 20 Jul 2007 22:25:41 +0000 (22:25 +0000)]
Add support for default "say mode" (whether to use the "old" method or "new" method. "new" method being config file)
Add support for autocomplete of "say load" CLI command.
When using users.conf for the entries in the directory, if multiple users had the same last name, only the first user listed would be available
in the directory.
(closes issue #10200, reported by mrskippy, patched by me)
Use the define that specifies the default length of an artificially created
DTMF digit in the ast_senddigit() function. The define is set to 100ms by
default, which is the same thing that this function was using. But, using
the define lets changes take effect in this case, as well as the others where
it was already used.
(closes issue #10247)
Reported by: fkasumovic
Patches:
chan_sip.patch uploaded by fkasumovic (license #101)
Drop any peer realm authentication entries when reloading so multiple entries do not get added to the peer.
(closes issue #10246)
Reported by: fkasumovic
Patches:
res_conver.patch uploaded by fkasumovic (license #101)
Use the last occurance of . to find the extension, not the first occurance.
Luigi Rizzo [Fri, 20 Jul 2007 14:54:01 +0000 (14:54 +0000)]
Extend the 'network settings' section with indication on the
localnet settings (requires the change in SVN 76034), and also
give an indication on whether/why/how the remapping of addresses
in SIP message is done or not.
I think this is especially useful for debugging the configuration,
as the address remapping depends on a combination of at least 3
parameters (localnet, externhost, externip) and successful DNS lookup.
An example of the output of this section is below:
I leave to the community the judgement if the above info is a
useful addition for 1.4. It is not a bugfix, but it is neither a
new feature, only a useful diagnostic tool.
Note that I would like to move there also the bindaddress/port
information, in the usual addr:port format e.g.
Luigi Rizzo [Fri, 20 Jul 2007 14:38:36 +0000 (14:38 +0000)]
expose struct ast_ha so external code can do things such as printing it
(e.g. chan_sip.c in a subsequent commit).
Obviously exposing the internals of a data structure is far from ideal
(especially in a case like this where the implementation is very
inefficient and will need to be changed at some point).
On the other hand, it was also unclear what additional APIs should
we provide instead, and because exposing the stucture has no impact
on source and binary compatibility, this seemed to me the best option at
this time.
Steve Murphy [Thu, 19 Jul 2007 23:24:27 +0000 (23:24 +0000)]
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
Changes in handling return values of several functions in app_queue. This all started as a fix for issue #10008
but now includes all of the following changes:
1. Simplifying the code to handle positive return values from ast API calls.
2. Removing the background_file function.
3. The fix for issue #10008
(closes issue #10008, reported and patched by dimas)
Russell Bryant [Thu, 19 Jul 2007 15:59:19 +0000 (15:59 +0000)]
(closes issue #10210, reported and patched by juggie)
This merges the trunk only part of the patches from this issue. In 1.4, res_agi
will issue a warning if you try to use DeadAGI on a channel that is not hung up.
Now, in trunk, it just plain won't let you do it.
When processing full frames, take sequence number wraparound into account when
deciding whether or not we need to request retransmissions by sending a VNAK.
This code could cause VNAKs to be sent erroneously in some cases, and to not
be sent in other cases when it should have been.
(closes issue #10237, reported and patched by mihai)
Luigi Rizzo [Thu, 19 Jul 2007 08:07:04 +0000 (08:07 +0000)]
print more of the network settings (externip, externhost etc.)
in the "sip show settings" cli output. I have put these in a
separate section, probably even bindaddr and SIP port should go
there.
There are more things to add here e.g. localnet and so on.
projects / thirdparty / asterisk.git / log
