external libraries and URLs to these. Please help me add these
references.
We might want to create a similar macro "\linuxpackage" to list
the needed Linux packages in popular distributions.
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See skinny.conf.sample for configuration example.
Note: Some devices (seen on 12SP+/30VIP) will lock up if they monitor too many hints.
This seems to be a hardware limitation - there isn't anything we can do about it.
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r56569 | qwell | 2007-02-23 20:02:53 -0600 (Fri, 23 Feb 2007) | 4 lines
Make sure to set a speeddials parent on creation.
Don't crash if hold is pressed when no call is active.
Don't return in places that we shouldn't..
Update softkey map when call is connected
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There is not a large amount of code here and the changes are not very invasive.
However, they should significantly improve performance of chan_iax2 under load.
IAX2 media frames only carry the *source* call number. So, when one arrives,
the correct session that it is a part of has to be matched on IP address, port
number, and call number, instead of just a call number. Had these frames
carried the *destination* call number, this would not be an issue, because that
would be a unique identifier that would make it easy to immediately identify
the correct session.
The way that chan_iax2 did this matching was extremely inefficient. It starts
at the first available call number and traverses each call number sequentially,
locking and unlocking a mutex for each one, to try to match against it. It
would do this regardless of whether the call number was in use or not. So,
for a call with a local call number of 25000, every single incoming media
frame would require a traversal that required 25000 mutex lock and unlock
operations. (Note that the max call number is about 32k).
I have introduced a hash table of active IAX2 calls to improve this lookup
process. The hash is done on the IP address, port number, and call number.
So, for the previous example, a few lock/unlock operations may be done versus
25000 for each frame.
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r56407 | russell | 2007-02-23 14:20:00 -0600 (Fri, 23 Feb 2007) | 12 lines
Merged revisions 56406 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r56406 | russell | 2007-02-23 14:17:56 -0600 (Fri, 23 Feb 2007) | 4 lines
Don't destroy mutexes before unregistering all of the entry points from the core.
Also, fix a potential memory leak from not destroying the locks for all of the
possible call numbers (about 32k of them).
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r55914 | file | 2007-02-21 12:18:19 -0500 (Wed, 21 Feb 2007) | 2 lines
Add a flag that indicates whether a SIP dialog is an outgoing call or not. SIP_OUTGOING originally did it but it was repurposed to the direction of the last transaction, which can cause update_call_counter to falsely decrease the wrong counters. (please don't hurt me oej) (issue #8943 reported by mdu113)
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r55397 | dbailey | 2007-02-19 08:52:59 -0600 (Mon, 19 Feb 2007) | 3 lines
Changed iax2 process thread to detached to correct memory leak due to left over thread context on thread exit.
Modified module unload process to avoid deadlocks on pthread cancels
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convert various #if expressions to #ifdef for macros that may not be defined (and where the value is not important)
Note: two of these changes are in bison generated files which is going to be inconvenient when they are regenerated
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r55002 | file | 2007-02-16 17:18:46 -0500 (Fri, 16 Feb 2007) | 10 lines
Merged revisions 54999 via svnmerge from
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r54999 | file | 2007-02-16 17:13:45 -0500 (Fri, 16 Feb 2007) | 2 lines
Do not send indications through ast_indicate in chan_agent but instead go directly to the technology. This way when indications are emulated they happen on the Agent channel and do not screw up formats on the channels. (issue #8439 reported by punkgode)
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T.140/RFC 2793 is a live communication channel, originally
created for IP based text phones for hearing impaired.
Feels very much like the old Unix talk application.
This code is developed and disclaimed by John Martin of Aupix, UK.
Tested for interoperability by myself and Omnitor in Sweden,
the company that wrote most of the specifications.
A big thank you to everyone involved in this.
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r54204 | russell | 2007-02-13 13:42:00 -0600 (Tue, 13 Feb 2007) | 5 lines
If we fail to create the SIP socket, then return -1 from reload_config() so
that load_module() will return AST_MODULE_LOAD_DECLINE. Otherwise, the console
will just get spammed with error messages every time chan_sip tries to send a
message.
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This replaces the older, broken, implementation where a setting in
[general] did not do anything and the [peer] part was broken.
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r53810 | russell | 2007-02-09 18:35:09 -0600 (Fri, 09 Feb 2007) | 24 lines
Merge team/russell/sla_rewrite
This is a completely new implementation of the SLA functionality introduced in
Asterisk 1.4. It is now functional and ready for testing. However, I will be
adding some additional features over the next week, as well.
For information on how to set this up, see configs/sla.conf.sample
and doc/sla.txt.
In addition to the changes in app_meetme.c for the SLA implementation itself,
this merge brings in various other changes:
chan_sip:
- Add the ability to indicate HOLD state in NOTIFY messages.
- Queue HOLD and UNHOLD control frames even if the channel is not bridged to
another channel.
linkedlists.h:
- Add support for rwlock based linked lists.
dial.c:
- Add the ability to run ast_dial_start() without a reference channel to
inherit information from.
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r53109 | oej | 2007-02-02 01:24:03 +0100 (Fri, 02 Feb 2007) | 4 lines
Disable the direct p2p RTP call setup in SIP. You can enable it in sip.conf, but it is now
considered experimental until we solve the AST_CONTROL_ANSWER with payload and videocaps
stuff.
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r53085 | oej | 2007-02-01 22:05:34 +0100 (Thu, 01 Feb 2007) | 4 lines
- Clean INC_COUNT flag when we decrement call counter
- If it's still set at time of dialog destruction, make sure we decrement the device call counter properly
before we destroy the dialog
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If you set call limit and busy limit, chan_sip will indicate BUSY for a device
that has reached the busy limit and allow calls up to the call limit, allowing
for call transfers (that generate a new call).
If you only set call limit, chan_sip will not indicate BUSY until that limit
is filled.
This affects SIP subscriptions, call queues and manager applications.
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r52952 | russell | 2007-01-30 13:33:12 -0600 (Tue, 30 Jan 2007) | 5 lines
Only set the DTMF flag on the rtp structure if the DTMF mode is actually
RFC2833, not just that it is not INFO. This makes it get set for inband DTMF
as well, which is not valid.
(issue #8936)
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r51328 | russell | 2007-01-19 13:08:25 -0600 (Fri, 19 Jan 2007) | 5 lines
Fix VLDTMF support in chan_gtalk. AST_FRAME_DTMF and AST_FRAME_DTMF_END are
actually the same thing. So, a digit would have been interpreted incorrectly
here. Since the channel driver will always have the begin and end callbacks
called for a digit, only support the button-down and button-up messages.
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r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines
Merge the changes from the /team/group/vldtmf_fixup branch.
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597, maybe others...)
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r51204 | russell | 2007-01-17 16:09:52 -0600 (Wed, 17 Jan 2007) | 4 lines
Instead of dividing the offset by 2 directly, make it more clear that the
offset is being scaled by the size of the elements in the buffer.
(Inspired by a discussing on the asterisk-dev list about this code)
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r51087 | file | 2007-01-16 00:55:23 -0500 (Tue, 16 Jan 2007) | 10 lines
Merged revisions 51085 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r51085 | file | 2007-01-16 00:53:31 -0500 (Tue, 16 Jan 2007) | 2 lines
Add none as a valid callgroup/pickupgroup option. I consider it a bug that it would inherit it all the way down and not have any way to reset it to nothing - so that's why it is in 1.2. (issue #8296 reported by gkloepfer)
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selectable by how it is called in the dialplan. This allows a speaker
system hooked up to the soundcard to be used for both ring notification,
as well as paging.
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r50468 | file | 2007-01-11 00:53:09 -0500 (Thu, 11 Jan 2007) | 2 lines
Remove check for channel state as it can definitely be something other then ring, and also clean up the code a bit. This should solve the parking issues and maybe some attended transfer issues people have been seeing.
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r50466 | file | 2007-01-11 00:19:39 -0500 (Thu, 11 Jan 2007) | 2 lines
Add support to see whether NAT was detected (yay symmetric RTP) and also add a check in chan_sip so that if NAT has been detected and the reinvite behind nat option has been turned off, then just do partial bridge. (issue #8655 reported by mnicholson)
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r50006 | oej | 2007-01-08 15:26:14 +0100 (Mon, 08 Jan 2007) | 11 lines
Issue #8677 - Handle failure of T.38 re-invite
This is not a fix, but adding an error message to tell the admin that
we have a bad configuration. We should not send T.38 re-invites to devices
that can't handle it (with the current architecture where you have to
hard-code t.38 support per device).
To really fix this, we need to figure out a way to tell the incoming
call that the re-invite failed, so we can signal failure on that
end and go back to the original call.
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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.
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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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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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Russell Bryant
Olle Johansson
Jason Parker
Joshua Colp
Doug Bailey
Kevin P. Fleming
Matthew Fredrickson
Paul Cadach
Christian Richter
Luigi Rizzo
Dwayne M. Hubbard
Tilghman Lesher
Matt O'Gorman
Steve Murphy
Nadi Sarrar