https://origsvn.digium.com/svn/asterisk/branches/1.4
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r49581 | kpfleming | 2007-01-04 17:50:15 -0600 (Thu, 04 Jan 2007) | 3 lines
create the IAX2 processing threads as background threads so they will use smaller stacks
when we create a dynamic thread, put it on the dynamic_list right away so we don't lose track of it
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r49457 | kpfleming | 2007-01-04 12:05:47 -0600 (Thu, 04 Jan 2007) | 2 lines
make building of codec_gsm against the system GSM library actually work
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r49460 | kpfleming | 2007-01-04 12:16:40 -0600 (Thu, 04 Jan 2007) | 2 lines
don't define this type either if LOW_MEMORY is enabled
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r49461 | kpfleming | 2007-01-04 12:17:01 -0600 (Thu, 04 Jan 2007) | 2 lines
don't do frame header caching in the core if LOW_MEMORY is defined
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r49313 | crichter | 2007-01-03 10:06:50 +0100 (Mi, 03 Jan 2007) | 41 lines
Merged revisions 48319,48321,48467,48552,48576,49135,49303 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) | 1 line
changed a few debugs to higher debug levels
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r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) | 1 line
added the export and import of the MISDN_ADDRESS_COMPLETE Variable to inidcate wether the extension is already completely dialed or if there might come additional digits by information elements. also added some docs for that.
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r48467 | crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line
removed FIXUP state. added check for channel allocation conflict when we create a setup while the other site creates a setup on the same channel, besides the check we resolve this conflict.
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r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18 Dez 2006) | 1 line
when our PTP Partner sends us a SETUP with a preselected channel we just accept it, even when we're NT. added some checks for segfaults.
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r48576 | crichter | 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line
when we reject a channel, because it's in use already, we shouldn't process the setup anymore. made the channel allocation a bit easier and more understandable, removed a few unused lines
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r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02 Jan 2007) | 1 line
added check for channel ranges in the set/empty channel functions. set pmp_l1_check default to no. added misdn restart pid cli command. added cleaning of channel when we send a RELEASE_COMPLETE.
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r49303 | crichter | 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines
* Added check for bridging in misdn_call to avoid setting echocancellation
when 2 mISDN channels are involved and when bridging is set. That lead
to a kernel panic before under different situations, because we switched
about 2 times between hardware bridging and echocancelation
* readded MISDN_URATE variable which got lost before, this should make app_v110
work again
* fixed typo
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r49259 | file | 2007-01-02 20:19:53 -0500 (Tue, 02 Jan 2007) | 2 lines
Check pvt structure presence before passing to send_command. This gets rid of the irritating message about a packet without pvt structure. This happens because the scheduled item is getting cancelled at almost the exact moment it is getting executed.
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r49102 | kpfleming | 2007-01-01 17:34:35 -0600 (Mon, 01 Jan 2007) | 2 lines
check specifically for VLDTMF and transcoding support in the system's Zaptel installation, and make only the modules that need those features dependent on them (this will allow building the other Zaptel-using parts of Asterisk against older versions of Zaptel or those on other platforms that haven't caught up yet to the Linux version)
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r48964 | file | 2006-12-25 23:31:58 -0500 (Mon, 25 Dec 2006) | 2 lines
Add an API call that initializes an RTP structure. We need this because chan_sip is cheeky and uses a temporary RTP structure for codec purposes, and the API calls that are used rely on the lock. (Pointed out on asterisk-dev by Andy Wang)
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defined in indications.h to ind_tone_zone_sound and ind_tone_zone,
to avoid conflicts with the structs with the same names
defined in tonezone.h
Hope i haven't missed any instance.
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r48564 | file | 2006-12-18 12:15:49 -0500 (Mon, 18 Dec 2006) | 2 lines
Put thread into proper list if we abort handling due to an error, and also hold the lock while putting it back into the proper idle list so we don't prematurely get a signal. (issue #8604 reported by arkadia)
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just failing to compile.
It seems like the proper way to do this would be in the configure script.
However, that wouldn't help existing checkouts unless we forced the configure
script to be executed after any code was changed.
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r48363 | russell | 2006-12-09 10:59:42 -0500 (Sat, 09 Dec 2006) | 8 lines
Use locking when accessing the registrations list. This list is not actually
used very often, so the likelihood of there being a problem is pretty small,
but still possible. For example, if the CLI command to list the registrations
was called at the same time that a reload was occurring and the registrations
list was getting destroyed and rebuilt, a crash could occur.
In passing, go ahead and convert this list to use the linked list macros.
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frames. The function, before this commit, was roughly 1400 lines long. So, I
am working on breaking this up into functions so that the code is easier to
follow and debug. Also, I will be committing these changes in chunks as I do
them to ease tracking down any potentially introduced problems.
Break out roughly 150 lines from socket_process() and introduce a new function,
socket_process_meta() which handles the parsing of an incoming meta frame.
Also, restructure some of this code a bit to reduce the deep nesting that was
in this code.
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- Use sizeof() to pass an array size to a function
- Use a single bit for a variable in the chan_iax2_pvt stuct since that is all
it needs.
- Add some comments about the iaxs, iaxl, and lastused arrays.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48359 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is imported from branch "invitestate" and "invitestate-1.4"
***
***
*** IF YOU HAVE ISSUES WITH BYEs/CANCELs - PLEASE UPDATE AND TEST AGAIN!
*** Thank you!
***
***
/Olle
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- Encapsulate RTP timers to the RTP structure, so we have one set for video and one for audio
- Document RTP keepalive configuration option
- Cleanup and document the monitor support function to hangup on RTP timeouts
- Add RTP keepalive to SIP show settings
Imported from 1.4 with modifications for trunk.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47823 | rizzo | 2006-11-18 18:59:35 +0100 (Sat, 18 Nov 2006) | 5 lines
fix bug 7450 - Parsing fails if From header contains angle brackets
(the bug was only in a corner case where the < was right after the
opening quote, and the fix is trivial).
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doing busy-wait on the output audio device.
As it is set now, it tries to push a frame every 10ms,
which is still too frequent but avoids deep restructuring
of the code (which i should do, though).
Note, this is only for ring tones, regular audio coming
from the network is still delivered as soon as it is
available.
Eventually this could well end up in the 1.4 branch, but
since i am probably the only user of chan_oss there isn't
much urgency to do that.
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clean platform for the handling of BYE/CANCEL than what we had. It might also need to changes
in other parts of the code, since we know the state of the INVITE transaction.
Please observe that this is is not dialog states at all, this is INVITE transaction states.
Hello Michael Proctor, and thank you! :-)
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and related code rearrangement.
Approved by: oej
This could be made better if we declared
struct sip_pvt *dialpg = pkt->owner;
at the beginning of the function, and use it throughout the function.
I'll let the boss decide :)
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simplify __sip_ack() removing a strcmp for looking up packets.
no functional change, only performance, so don't need to merging
to earlier branches now.
Approved By: oej
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versions of the format string are identical. Also, since each format is only
used once, get rid of the use of defines all together. (issue #8344, julieng)
In passing, also clean up the formatting a but to get rid of the nesting
without the use of braces, as defined in the coding guidelines.
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- Keep RTP running during T.38 session
We might improve the code to issue ast_rtp_stop if T.38 re-invite not fails
later on in the code, but I don't see many reasons to.
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r47391 | russell | 2006-11-09 16:26:27 -0500 (Thu, 09 Nov 2006) | 7 lines
Work around an issue that caused menuselect to display a bogus description for
app_voicemail and chan_zap. These modules use some preprocessor directives to
determine what it will report to Asterisk as its description. However, the way
we extract this information from the source files for menuselect is not smart
enough to figure this out.
(issue #8326, #8328)
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r47380 | file | 2006-11-09 11:53:25 -0500 (Thu, 09 Nov 2006) | 10 lines
Merged revisions 47379 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r47379 | file | 2006-11-09 11:48:05 -0500 (Thu, 09 Nov 2006) | 2 lines
Don't include compiler.h on kernels 2.6.18 and higher as, well, it's apparently going to be removed. This should make all you FC6 fans happy as your Asterisk will now build without any mods.
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- Fix documentation for sip_pvt_lock/unlock - doxygen does not inherit like zapata.conf !!!
- Change doc for a sip_pvt setting
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avoid doing p > 0 when p is a pointer;
move a lock closer to the place where it is needed
Approved By: oej
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Same as for peers and users, replace ASTOBJ_UNREF(r, sip_registry_destroy)
with unref_registry(r);
Approved By: oej
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Replace ASTOBJ_UNREF(peer, sip_destroy_peer) with unref_peer(peer);
This places the name of the destructor in one place only (where it
should be), eliminates the chance of errors in case you specify the wrong
destructor, and also lets the compiler do type checking on the argument,
again helping with keeping the code clean.
Same for users.
remove two duplicate definitions.
Approved By: oej
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Overall i think the previous change to ast_channel_alloc()
to close bug 7506 should have been done by defining
an ast_set_callerid_noevent() function that does the
setting without generating the event.
Lot less code duplication, and easier to handle.
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to the peer side of a type=friend.
This is for trying to support BJ in his quest to solve some issues
with the queue system and type=friend objects.
BJ: Please test!
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I think this module doesn't compile, anyways, because
it has not been updated to the new module interface.
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e.g. in the Manager interface. This information is available as
a callerid (or something like that) during a call, but not when a
device is registered but silent.
It may be useful to have it available e.g. when developing a user
interface/operator panel, to map numbers to names.
experimental, so not committed to 1.4
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r46775 | file | 2006-11-01 13:21:34 -0500 (Wed, 01 Nov 2006) | 2 lines
It's another round of chan_iax2 fixes! Should hopefully fix the deadlock issues people have been reporting. IAXtel now has qualify turned on for 800 peers and it is handling it fine.
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is to the person that configures asterisk. That we use it internally in the
contact header is a totally different story.
Still not convinced this is a good option.
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- removing transmit_reinvite_with_t38_sdp in favour of adding an argument to
transmit_reinvite_with_sdp
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lock when needed - when we remove the dialog from the dialog list
If this doesn't lead to severe problems, it might help with some locking issues
in 1.4/1.2.
- Remove the term "interface" as a synonym for a SIP dialog. Sorry, Mark, but no
one understands it... ;-)
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r46358 | russell | 2006-10-27 10:32:40 -0500 (Fri, 27 Oct 2006) | 5 lines
Instead of iterating all of the options once to look for jitterbuffer options,
and then again for everything else, move the processing of jitterbuffer
options into the main loop so that there are no erroneous messages about
ignoring unknown options. (issue #8226)
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r46351 | crichter | 2006-10-27 11:49:20 +0200 (Fr, 27 Okt 2006) | 9 lines
Merged revisions 46176 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r46176 | crichter | 2006-10-25 10:41:59 +0200 (Mi, 25 Okt 2006) | 1 line
added nttimeout option to configure wether we disconnect calls on NT timeouts or not during an overlapdial session
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r46352 | crichter | 2006-10-27 11:58:44 +0200 (Fr, 27 Okt 2006) | 1 line
fixed not compile issue, which was just introduced
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r46353 | crichter | 2006-10-27 12:03:23 +0200 (Fr, 27 Okt 2006) | 9 lines
Merged revisions 46350 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r46350 | crichter | 2006-10-27 11:24:01 +0200 (Fr, 27 Okt 2006) | 1 line
fixed a bug which caused chan_misdn to try to allocate 2 times the same channel on high load, which then caused instability of mISDN. removed a useless function from isdn_lib.c
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r46082 | kpfleming | 2006-10-23 22:45:42 -0500 (Mon, 23 Oct 2006) | 2 lines
add an API call to allow channel drivers to determine which media formats are compatible (passthrough or transcode) with the format an existing channel is already using
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r46083 | kpfleming | 2006-10-23 22:53:32 -0500 (Mon, 23 Oct 2006) | 2 lines
ensure that the translation matrix is properly lock-protected every place it is used
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r46152 | kpfleming | 2006-10-24 18:45:19 -0500 (Tue, 24 Oct 2006) | 2 lines
if multiple translators are registered for the same source/dest combination, ensure that the lowest-cost one is always inserted earlier in the list
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r46153 | kpfleming | 2006-10-24 19:10:54 -0500 (Tue, 24 Oct 2006) | 2 lines
code zone experiment: don't offer formats in the outbound INVITE that aren't either passthrough or translatable
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when the number of channels fill the MTU on a given link.
In the future, this needs to be configurable per peer with trunking enabled.
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appreciated really. (Read the coding guidelines).
I've worked hard to make chan_sip a better place to code in, let's
keep it that way and don't add more stuff without comments.
Thank you.
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so fix the places where this might happen.
This is also a fix that ought to go into 1.4
[The difference between the two functions is a bit confusing,
and in asterisk i believe all string handling functions
should be able to handl a NULL string as argument,
but changing the API in trunk and not in 1.4 would make
backporting harder.]
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As discussed on the mailing lists, 0 is a legal value
for Cseq, so there is no point to treat it specially.
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for NULL is certainly wrong and usually disables the
checks that we want to make instead.
This commit fixes a number of the above bugs where the result
of get_header() is immediately checked for NULL.
This is certainly a candidate for merging into 1.4
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On passing, remove a wrong comment (that probably I wrote
myself!) and introduce a temporary variable to avoid a
misleading cast.
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simplifies its usage.
+ add another client for parse_uri, in handling Contact: strings
(on passing, document the content of the "fullcontact" field);
+ in register_verify(), mark with XXX what i believe is another
misinterpretation on the URI format when '@' is missing.
No code changed here, so no fixes applied.
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fix parse_uri() to interpret a missing userinfo
section as a domain-only URI, and comment a wrong
interpretation of the above in check_user_full().
The function has been patched to preserve the existing
behaviour (in what admittedly is a corner case, but
could be received under attacks).
Hopefully the From: based matching will go away soon!
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before splitting around the @, otherwise the refer_to_domain
might contain arguments as well, causing failures.
I think this is a true bug that ought to be fixed in 1.4 as well.
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introducing the function parse_uri() that splits
a URI in its components.
Right now use it only in one place, because the custom
parsing that is done here and there sometimes has
bugs that i want to figure out first.
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apart from a small (but disabled by default) new option.
In detail:
+ introduce a new value for enum check_auth_result, AUTH_DONT_KNOW,
used (read below) when a function does not have a conclusive response.
Possibly this is the same as AUTH_NOT_FOUND, but need to check further.
+ move the large blocks (checking in the users list and in the peers
list, respectively) from check_user_full() to separate functions.
They return AUTH_DONT_KNOW in case they don't find a match, so
the caller know that it has to try the next method.
There is still some duplication of code here, but i
have not tried yet to remove it.
+ [new option] a new option in sip.conf, match_auth_username,
has been introduced, and disabled by default.
If set, and the incoming request carries authentication info,
the username to match in the users list is taken from there
rather than from the From: field.
This change is easy to identify, being made of
- one line to declare the variable match_auth_username
- a block of 15 lines in check_user_full()
- one line in sip list settings
- two lines for parsing the config file.
check_user_full() is now a lot cleaner - basically a sequence of
checks that are applied to the request. This will help future
work with new matching schemes.
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A large block needs reindentation now, but we don't do that because
it can be moved to a separate function.
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lock these data structures.
This improve readability, and also hides the underlying
locking mechanism so it is a lot easier to add diagnostic
code, or move the object locks somewhere else, etc.
On passing, rename the lock field in sip_pvt to pvt_lock,
also for ease of readability.
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allow custom threadstorage init functions to return failure
use a custom init function for chan_sip's temp_pvt, to improve performance a bit
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be called for each thread specific object after they are allocated. Note that
there was already the ability to define a custom cleanup function. Also, if
the custom cleanup function is used, it *MUST* call free on the thread
specific object at the end. There is no way to have this magically done that
I can think of because the cleanup function registered with the pthread
implementation will only call the function back with a pointer to the
thread specific object, not the parent ast_threadstorage object.
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r45408 | kpfleming | 2006-10-17 17:24:10 -0500 (Tue, 17 Oct 2006) | 3 lines
optimize the 'quick response' code a bit more... no more malloc() or memset() for each response
expand stringfields API a bit to allow reusing the stringfield pool on a structure when needed, and remove some unnecessary code when the structure was being freed
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r45378 | file | 2006-10-17 16:30:34 -0400 (Tue, 17 Oct 2006) | 2 lines
Don't create a "real" pvt structure for requests that shouldn't be able to create one. Instead use a temporary pvt and fill it with enough information so we can send a reply.
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I *know* it is not required, but it makes navigation easier and will help
when splitting up this large source code file.
Thank you!
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necessary - rather, cast the argument to int.
In this case, the string is in a UDP packet and as such
limited to 64k so its length can be safely represented in an int
without truncation (besides, this is just a debugging message!)
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To avoid the same mistake in the future (due to slightly
confusing variable names), add a comment.
On passing, remove a redundant initialization.
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and friends in a function, auth_headers(), which is used to
simplify the interface of do_{proxy|register}_auth().
+ use PROXY_AUTH = 407, WWW_AUTH = 401 as values for enum sip_auth_type;
No functional change, only code cleanup.
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authentication issues. This was committed in revision 44844, where the commit
message was just "small formatting cleanup", so I am pretty sure he didn't mean
to commit this part.
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r44561 | crichter | 2006-10-06 14:50:25 +0200 (Fr, 06 Okt 2006) | 9 lines
Merged revisions 44334 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r44334 | crichter | 2006-10-04 17:13:58 +0200 (Mi, 04 Okt 2006) | 1 line
added the option 'reject_cause' to make it possible to set the RELEASE_COMPLETE - cause on the 3. incoming PMP channel, which is automatically rejected because chan_misdn does not support that kind of callwaiting. Therefore chan_misdn supports now 3 incoming channels on a PMP BRI Port. misdn_lib_get_free_bc now gets the info if the requested channel is incoming or outgoing to make the 3. channel possible
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and rename the old DEFAULT_SIP_PORT as STANDARD_SIP_PORT
to make it clear that this is not something we can change,
unlike other defaults.
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that, but the protocol clearly states that if we DO NOT mention a port it
is 5060. DEFAULT_SIP_PORT is whatever we default to listen to.
I believe it's the third time I revert a patch like this.
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1. slightly rearrange/simplify the parsing of the argument in sip_register.
This brings in a patch that has been in Mantis (5834) for ages,
and is the larger part of the commit;
2. implement the "contact" option for peers, similar to the one in users.conf:
If you put a "contact" option with a non-empty argument (e.g. contact=123)
in a peer section, asterisk will register with the provider as if you had a
register= username:secret@host/contact
line in the general section.
The latter is a very small is a new feature so i am not putting it
in the 1.4 branch, although the "contact" option in user.conf is
already in the 1.4 branch and so it wouldn't be too strange to
merge it.
Note that the implementation of "contact" is much simpler than
the one in 5834, and limited to a few lines in build_peer().
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r44378 | kpfleming | 2006-10-04 14:47:22 -0500 (Wed, 04 Oct 2006) | 4 lines
update thread creation code a bit
reduce standard thread stack size slightly to allow the pthreads library to allocate the stack+data and not overflow a power-of-2 allocation in the kernel and waste memory/address space
add a new stack size for 'background' threads (those that don't handle PBX calls) when LOW_MEMORY is defined
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r44078 | pcadach | 2006-09-30 22:12:23 +0600 (Сбт, 30 Сен 2006) | 6 lines
Fix issue #7928 correctly. Next is a comment of previous fix:
Issue #7928 - Don't send both 404 and 503. Fix by phsultan with
a small fix by me, myself or I. Thanks, Philippe!
(This was caused by my changes to the transaction handling)
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r44068 | pcadach | 2006-09-30 10:37:39 +0600 (Сбт, 30 Сен 2006) | 14 lines
Found some buggy SIP clients (phones Planet VIP-153T firmware
1.0, Linksys PAP2 firmware 3.1.9(LSc)) which sends ACK not on OK
message only (when remote party answers) but on RINGING message
too, so when we send 200 OK message, we get unidentified ACK
message (because INVITE acknowledged on RINGING message already),
so 200 OK retransmits within its retransmission interval then
call gets dropped.
If someone else knows how to provide workaround for such cases,
please, fix it in correct way.
Thanks to ssh from #asteriskru for provide access to his box to
study and fix this case.
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r44034 | pcadach | 2006-09-30 02:43:13 +0600 (Сбт, 30 Сен 2006) | 1 line
Fake display name by called number on incoming calls (until passing connected number/connected name is not implemented)
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r44042 | pcadach | 2006-09-30 03:05:43 +0600 (Сбт, 30 Сен 2006) | 1 line
Set TON/PRESENTATION information more carefully when no CallingNumber IE available
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r44043 | pcadach | 2006-09-30 03:09:10 +0600 (Сбт, 30 Сен 2006) | 1 line
Compile first, please
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r43861 | pcadach | 2006-09-28 18:47:23 +0600 (Чтв, 28 Сен 2006) | 1 line
Put attribute tag at correct place
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r43862 | pcadach | 2006-09-28 18:58:22 +0600 (Чтв, 28 Сен 2006) | 1 line
Force remote side to start media on outgoing PROGRESS message
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r43635 | pcadach | 2006-09-26 03:26:12 +0600 (Втр, 26 Сен 2006) | 1 line
Fix ASN1 description of non-standard Cisco extensions
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r43843 | pcadach | 2006-09-28 12:01:37 +0600 (Чтв, 28 Сен 2006) | 1 line
Don't treat unknown control frames as voice
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r43844 | pcadach | 2006-09-28 12:02:45 +0600 (Чтв, 28 Сен 2006) | 1 line
Don't warn on HOLD/UNHOLD control frames
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r43846 | pcadach | 2006-09-28 16:51:21 +0600 (Чтв, 28 Сен 2006) | 1 line
Do not open transmit channel until TCS is received
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r43783 | file | 2006-09-27 13:00:31 -0400 (Wed, 27 Sep 2006) | 2 lines
Get rid of two functions from a time now past (we THINK these are from pre-recursive lock time) that may be contributing to two open issues on the bug tracker (7562/7939) and that has the potential to just make bad things happen if the timing is right.
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r43650 | qwell | 2006-09-26 08:33:47 -0700 (Tue, 26 Sep 2006) | 11 lines
Add proper codec support to chan_skinny. Works with at least ulaw, alaw, and g729a.
This is technically a "new feature", but there are justifications for it.
I found a bug with the recent rtp packetization changes, which caused the media setup to
fail under certain circumstances, particularly when using allow=all, or having no allow=
statements (globally or on the device).
I could have either removed the rtp packetization features, or I could add proper codec
support (which, without, I think most people would consider to be a bug anyways).
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r43564 | russell | 2006-09-24 10:58:10 -0400 (Sun, 24 Sep 2006) | 5 lines
Fix a CLI command registration issue where an erroneous message claiming that
"iax2 show provisioning" was already registered. This was because this command
was registering itself as both the command, as well as the command it is
deprecating. (issue #8022, reported by bjweeks, fixed by myself)
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r43553 | russell | 2006-09-24 09:53:35 -0400 (Sun, 24 Sep 2006) | 12 lines
Merged revisions 43552 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r43552 | russell | 2006-09-24 09:50:30 -0400 (Sun, 24 Sep 2006) | 4 lines
Check to see if the channel that is activating the IAXPEER function is actually
an IAX2 channel before proceeding to process it to avoid crashing.
(issue #8017, reported by admott, fixed by myself)
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r42133 | bweschke | 2006-09-06 14:16:41 -0400 (Wed, 06 Sep 2006) | 6 lines
Look ma! No more deadlocks! <sic>
As posted from #7458 and others similar to it in Mantis:
p->app_lock was a mutex really designed for use with agents not in callback mode. That being the case, I've tried to code it so that when callback mode is used, the app_lock mutex will not be locked/unlocked at all. Please let me know how you make out - and if you continue to deadlock now, please reproduce the deadlock logging information and post to Mantis.
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changes in both of the moving specs. Currently chan_gtalk is
compatible with the latest gtalk/libjingle version, and chan_jingle
needs a lot of work.
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If this works properly, we might have to check 1.2 to implement a backport.
The theory is that if you get a final reply in a session, it is ok to destroy the session.
If you send a final reply, you need to keep the session open for potential retransmits
from the other side. If you send a HANGUP/CANCEL, wait to the other side confirms
or until you have a timeout. If you send HANGUP/CANCEL/ACK reliably, don't destroy
the session so that you cancel the needed retransmits.
I will have to change the timer to 64*T1, but that will be a separate patch. That will
mean that if we know the roundtrip time, we can destroy sessions quicker.
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now reports AST_MODULE_LOAD_DECLINE when loading if config file
is not there, also fixed an error in res_config_pgsql where it
had a non static function when it should.
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having a bunch of erroneous messages about attempting to double free frames
spammed to the console. Problem reported to me by file ...
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There are some situations in Asterisk where ast_frame and/or iax_frame
structures are rapidly allocatted and freed (at least 50 times per second
for one call).
This code significantly improves the performance of ast_frame_header_new(),
ast_frdup(), ast_frfree(), iax_frame_new(), and iax_frame_free() by keeping
a thread-local cache of these structures and using frames from the cache
whenever possible instead of calling malloc/free every time.
This commit also converts the ast_frame and iax_frame structures to use the
linked list macros.
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- restructured build tree and makefiles to eliminate recursion problems
- support for embedded modules
- support for static builds
- simpler cross-compilation support
- simpler module/loader interface (no exported symbols)
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channel to remove the theoretical race condition that the channel could get
bridged before the channel's jitterbuffer gets configured. This was pointed
out by PCadach on IRC. Thanks!
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* fail on misdn_cfg_init() if elements in the config enum don't match with the config structs in misdn_config.c
* implemented first bits for encoding ISDN facility information elements via ASN.1 descriptions
* using unnamed semaphore for syncing in misdn_thread
* advanced fax detection: configurable detect timeout and context to jump into
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* removed the holded element from the chan_list struct, we know this from the
state already
* added a few tweaks to make HOLD/RETRIEVE work again (TRANSFER does not work
yet)
* added possibility to debug mISDN frames via syslog
* added misdn_lib_port_is_blocked function to check if a port is blocked
* removed ec_training=1 from empty_bc, we don't use ec_training anymore
* removed unused misdn_lib_get_l2_status function
* added the nt bit to dummy misdn_bchannel objects
* setting bc->out_fac_type to FACILITY_NONE defaultly
* removed HANDLER_DEBUG stuff for better readability
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- instead of defining a free() wrapper in a bunch of files, define it as
ast_free() in utils.h and remove the copies from all the files.
- centralize and abstract the code used for doing thread storage. The code
lives in threadstorage.h, with one function being implemented in utils.c.
This new API includes generic thread storage as well as special functions
for handling thread local dynamic length string buffers.
- update ast_inet_ntoa() to use the new threadstorage API
- update ast_state2str() to use the new threadstorage API
- update ast_cli() to use the new threadstorage API
- Modify manager_event() to use thread storage. Instead of using a buffer of
4096 characters as the workspace for building the manager event, use a thread
local dynamic string. Now there is no length limitation on the length of the
body of a manager event.
- Significantly simplify the handling of ast_verbose() ...
- Instead of using a static char buffer and a lock to make sure only one
thread can be using ast_verbose() at a time, use a thread local dynamic
string as the workspace for preparing the verbose message. Instead of
locking around the entire function, the only locking done now is when the
message has been built and is being deliviered to the list of registered
verbose message handlers.
- This function was doing a strdup() on every message passed to it and
keeping a queue of the last 200 messages in memory. This has been
completely removed. The only place this was used was that if there were
any messages in the verbose queue when a verbose handler was registered,
all of the messages in the queue would be fed to it. So, I just made sure
that the console verbose handler and the network verbose handler (for
remote asterisk consoles) were registered before any verbose messages.
pbx_gtkconsole and pbx_kdeconsole will now lose a few verbose messages at
startup, but I didn't feel the performance hit of this message queue was
worth saving the initial verbose output for these very rarely used modules.
- I have removed the last three arguments to the verbose handlers, leaving
only the string itself because they aren't needed anymore. For example,
ast_verbose had some logic for telling the verbose handler to add
a newline if the buffer was completely full. Now that the buffer can grow
as needed, this doesn't matter anymore.
- remove unused function, ast_verbose_dmesg() which was to dispatch the
message queue
- Convert the list of verbose handlers to use the linked list macros.
- add missing newline characters to a few ast_verbose() calls
- convert the list of log channels to use the linked list macros in logger.c
- fix close_logger() to close all of the files it opened for logging
- update ast_log() to use a thread local dynamic string for its workspace
for preparing log messages instead of a buffer of size BUFSIZ (8kB on my
system) allocated on the stack. The dynamic string in this case is limited
to only growing to a maximum size of BUFSIZ.
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r39081 | russell | 2006-08-06 21:28:29 -0400 (Sun, 06 Aug 2006) | 7 lines
Fix a crash reported to me by hads on IRC. This crash would occur with the use
of the "distinctiveringaftercid" option. Also, on this user's system, the crash
would only occur when built without optimizations. This is because the bug is
that the code would write past the end of an array that was allocated on the
stack, and the structure of the stack is different with or without optimizations
enabled.
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- use appropriate types in some assignments
- use ast_strlen_zero()
- don't manually free cid fields since ast_set_callerid() will handle it
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r38903 | russell | 2006-08-05 01:07:39 -0400 (Sat, 05 Aug 2006) | 2 lines
suppress a compiler warning about the usage of a potentially uninitialized variable
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r38904 | russell | 2006-08-05 01:08:50 -0400 (Sat, 05 Aug 2006) | 10 lines
Fix an issue that would cause a NewCallerID manager event to be generated
before the channel's NewChannel event. This was due to a somewhat recent
change that included using ast_set_callerid() where it wasn't before. This
function should not be used in the channel driver "new" functions.
(issue #7654, fixed by me)
Also, fix a couple minor bugs in usecount handling. chan_iax2 could have
increased the usecount but then returned an error. The place where chan_sip
increased the usecount did not call ast_update_usecount()
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* added blocking flag to stack object. A port can be blocked/unblocked from the
cli
* added EVENT_PORT_ALARM to send alarm infos to the chan_misdn.c layer (later
we can add a manager event for that)
* added block_on_alarm option, to block the port whenever a ALARM occurs
* added need_busy flag to indicate if we've sended a CONTROL_BUSY already
* changed a bunch of cb_log(-1,..) to cb_log(0,..) due to funny behaviour in
recent asterisk ast_log messages..
* fixed a few ETSI state violations, especially when finishing calls in
different seldom states
* changed debug levels a lot to make the log more readable in low debuglevels
* some first fixes for the HOLD/RETRIEVE stuff (doesn't work totally still)
* removed the PRECONNECTED state stuff
* added cause 27 when we get a CLEANUP directly after a outgoing SETUP, this
creates a CHANISUNAVAIL instead of a NOANSWER
* removed the addr pointer from "misdn show stacks" that's not needed anymore
and makes the output more unreadable
* added cause saving on RELEASE/RELEASE_COMPLETE
* set cause to 16 on prepare_bc
* removed stack getting from ph_control functions, we don't really need it
there
* added beroec api
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r38611 | kpfleming | 2006-07-31 16:14:11 -0500 (Mon, 31 Jul 2006) | 4 lines
don't reissue hangup requests for SIP channels that have expired their RTP timeouts (one time is enough)
don't rescan the SIP private structure list too fast, it can cause channels to not be able to hang up (issue #7495, and probably others)
use ast_softhangup_nolock() since we already hold the channel's lock
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Kevin P. Fleming
Joshua Colp
Matt O'Gorman
Christian Richter
Olle Johansson
Russell Bryant
Luigi Rizzo
Jason Parker
Steve Murphy
Matthew Fredrickson
Tilghman Lesher
Nadi Sarrar
Paul Cadach
Mark Spencer
BJ Weschke