https://origsvn.digium.com/svn/asterisk/branches/10
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r331265 | rmudgett | 2011-08-09 18:12:49 -0500 (Tue, 09 Aug 2011) | 22 lines
Merged revisions 331248 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r331248 | rmudgett | 2011-08-09 17:12:59 -0500 (Tue, 09 Aug 2011) | 15 lines
Misc minor items found in code.
* Add some reentrancy protection in pbx.c when creating the contexts_table
hash table.
* Fix inverted test in chan_sip.c conditional code.
* Fix uninitialized variable and use of the wrong variable in chan_iax2.c.
* Fix test of return value in app_parkandannounce.c. Explicitly testing
for -1 is bad if the function does not actually return that value when it
fails.
* Fixup some comments and add some curly braces in features.c.
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r330586 | dvossel | 2011-08-02 11:17:59 -0500 (Tue, 02 Aug 2011) | 15 lines
Merged revisions 330581 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r330581 | dvossel | 2011-08-02 11:15:08 -0500 (Tue, 02 Aug 2011) | 8 lines
Fixes crash in chan_iax2.
Fixes crash in chan_iax2 resulting from an edge case in the
way control frames are queued during calltoken negotiation is complete.
(closes issue ASTERISK-17610)
Reported by: mgrobecker
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There is a fairly common pattern making its way through the code base where we
put a temporary object on the stack so we can call ao2_find() with OBJ_POINTER.
The purpose is so that it can be passed into the object hash function.
However, this really seems like a hack and potentially error prone. This patch
is a first stab at approach to avoid having to do that.
It adds a new flag, OBJ_KEY, which can be used instead of OBJ_POINTER in these
situations. Then, the hash function can know whether it was given an object or
some custom data to hash.
The patch also changes some uses of ao2_find() for iax2_user and iax2_peer
objects to reflect how OBJ_KEY would be used.
So long, and thanks for all the fish.
Review: https://reviewboard.asterisk.org/r/1184/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@330273 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r326411 | tilghman | 2011-07-05 17:08:29 -0500 (Tue, 05 Jul 2011) | 14 lines
Add the attribute "type" to each "<use>" for menuselect.
This matters only when autoconf fails to detect that weak linking is supported.
External optional dependencies will become optional in both cases, as they are
removed at compile time when not detected. However, runtime-optional modules
are made mandatory when weak linking is not found. This change affects only
the external optional dependencies; previously, they were incorrectly required
when weak linking support was not detected.
Patches:
20110702__issue18062__asterisk_trunk.diff.txt by tilghman (License #5003)
Tested by: iasgoscouk
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326412 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch is the foundation of an entire new way of looking at media in Asterisk.
The code present in this patch is everything required to complete phase1 of my
Media Architecture proposal. For more information about this project visit the link below.
https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal
The primary function of this patch is to convert all the usages of format
bitfields in Asterisk to use the new format and format_cap APIs. Functionally
no change in behavior should be present in this patch. Thanks to twilson
and russell for all the time they spent reviewing these changes.
Review: https://reviewboard.asterisk.org/r/1083/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Previously, I had added the ast_sched_thread stuff that was a generic scheduler
thread implementation. However, if you used it, it required using different
functions for modifying scheduler contents. This patch reworks how this is
done and just allows you to optionally start a thread on the original scheduler
context structure that has always been there. This makes it trivial to switch
to the generic scheduler thread implementation without having to touch any of
the other code that adds or removes scheduler entries.
In passing, I made some naming tweaks to add ast_ prefixes where they were not
there before.
Review: https://reviewboard.asterisk.org/r/1007/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299091 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r288194 | rmudgett | 2010-09-21 19:06:21 -0500 (Tue, 21 Sep 2010) | 40 lines
Merged revisions 288193 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r288193 | rmudgett | 2010-09-21 19:03:37 -0500 (Tue, 21 Sep 2010) | 33 lines
Merged revisions 288192 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r288192 | rmudgett | 2010-09-21 18:55:58 -0500 (Tue, 21 Sep 2010) | 26 lines
In chan_iax2.c:schedule_delivery() calls ast_bridged_channel() on an unlocked channel.
Near the beginning of schedule_delivery(), ast_bridged_channel() is called
on iaxs[fr->callno]->owner. However, the channel is not locked, which can
result in ast_bridged_channel() crashing should owner->tech change to a
technology that doesn't implement bridged_channel.
I also fixed the other calls to ast_bridged_channel() in chan_iax2.c since
the owner lock was not held there either.
Converted the existing channel deadlock avoidance to use
iax2_lock_owner(). Using the new function simplified some awkward code.
In the process of fixing the locking on ast_bridged_channel(), I also
found a memory leak in socket_process() for v1.6.2 and v1.8. The local
struct variable ies.vars is not freed on early/abnormal function exits.
(closes issue #17919)
Reported by: rain
Patches:
issue17919_v1.4.patch uploaded by rmudgett (license 664)
issue17919_w_leak_v1.6.2.patch uploaded by rmudgett (license 664)
issue17919_w_leak_v1.8.patch uploaded by rmudgett (license 664)
Review: https://reviewboard.asterisk.org/r/926/
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r286931 | jpeeler | 2010-09-15 14:22:15 -0500 (Wed, 15 Sep 2010) | 16 lines
Add parking extension for non-default parking lots.
This is a new feature that allows for parking to custom parking lots to be
accessed directly, rather than with channel variables or by changing the
default parking lot. The extension is set with the parkext option just as the
default parking lot is done. Also, the manager action has been updated to
optionally allow a specified parking lot.
(closes issue #14882)
Reported by: vmikhnevych
Patches:
patch_14882.txt uploaded by mnick (license 874)
modified by me
Review: https://reviewboard.asterisk.org/r/884/
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ACLs can now be configured to match IPv6 networks. This is only
relevant for ACLs in chan_sip for now since other channel drivers
do not support IPv6 addressing. However, once those channel drivers
are outfitted to support IPv6 addressing, the ACLs will already be
ready for IPv6 support.
https://reviewboard.asterisk.org/r/791
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277814 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The purpose of this patch is to eliminate struct ast_callerid since it has
turned into a miscellaneous collection of various party information.
Eliminate struct ast_callerid and replace it with the following struct
organization:
struct ast_party_name {
char *str;
int char_set;
int presentation;
unsigned char valid;
};
struct ast_party_number {
char *str;
int plan;
int presentation;
unsigned char valid;
};
struct ast_party_subaddress {
char *str;
int type;
unsigned char odd_even_indicator;
unsigned char valid;
};
struct ast_party_id {
struct ast_party_name name;
struct ast_party_number number;
struct ast_party_subaddress subaddress;
char *tag;
};
struct ast_party_dialed {
struct {
char *str;
int plan;
} number;
struct ast_party_subaddress subaddress;
int transit_network_select;
};
struct ast_party_caller {
struct ast_party_id id;
char *ani;
int ani2;
};
The new organization adds some new information as well.
* The party name and number now have their own presentation value that can
be manipulated independently. ISDN supplies the presentation value for
the name and number at different times with the possibility that they
could be different.
* The party name and number now have a valid flag. Before this change the
name or number string could be empty if the presentation were restricted.
Most channel drivers assume that the name or number is then simply not
available instead of indicating that the name or number was restricted.
* The party name now has a character set value. SIP and Q.SIG have the
ability to indicate what character set a name string is using so it could
be presented properly.
* The dialed party now has a numbering plan value that could be useful to
have available.
The various channel drivers will need to be updated to support the new
core features as needed. They have simply been converted to supply
current functionality at this time.
The following items of note were either corrected or enhanced:
* The CONNECTEDLINE() and REDIRECTING() dialplan functions were
consolidated into func_callerid.c to share party id handling code.
* CALLERPRES() is now deprecated because the name and number have their
own presentation values.
* Fixed app_alarmreceiver.c write_metadata(). The workstring[] could
contain garbage. It also can only contain the caller id number so using
ast_callerid_parse() on it is silly. There was also a typo in the
CALLERNAME if test.
* Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id
number string. ast_callerid_parse() alters the given buffer which in this
case is the channel's caller id number string. Then using
ast_shrink_phone_number() could alter it even more.
* Fixed caller ID name and number memory leak in chan_usbradio.c.
* Fixed uninitialized char arrays cid_num[] and cid_name[] in
sig_analog.c.
* Protected access to a caller channel with lock in chan_sip.c.
* Clarified intent of code in app_meetme.c sla_ring_station() and
dial_trunk(). Also made save all caller ID data instead of just the name
and number strings.
* Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge()
function.
* Corrected some weirdness with app_privacy.c's use of caller
presentation.
Review: https://reviewboard.asterisk.org/r/702/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds a generic API for accommodating IPv6 and IPv4 addresses
within Asterisk. While many files have been updated to make use of the
API, chan_sip and the RTP code are the files which actually support
IPv6 addresses at the time of this commit. The way has been paved for
easier upgrading for other files in the near future, though.
Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne
for their hard work on this.
(closes issue #17565)
Reported by: russell
Patches:
asteriskv6-test-report.pdf uploaded by russell (license 2)
Review: https://reviewboard.asterisk.org/r/743
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274783 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The external test suite stops Asterisk using the "core stop gracefully" command.
The logs from the tests show that there are a number of problems with Asterisk
trying to cleanly shut down. This patch addresses the following type of error
that comes from chan_iax2:
[Jun 22 16:58:11] ERROR[29884]: lock.c:129 __ast_pthread_mutex_destroy:
chan_iax2.c line 11371 (iax2_process_thread_cleanup):
Error destroying mutex &thread->lock: Device or resource busy
For an example in the context of a build, see:
http://bamboo.asterisk.org/browse/AST-TRUNK-739/log
The primary purpose of this patch is to change the thread pool shutdown
procedure to be more explicit to ensure that the thread exits from a point
where it is not holding a lock. While testing that, I encountered various
crashes due to the order of operations in unload_module() being problematic.
I reordered some things there, as well.
Review: https://reviewboard.asterisk.org/r/736/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272370 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r270866 | dvossel | 2010-06-16 12:35:29 -0500 (Wed, 16 Jun 2010) | 22 lines
fixes chan_iax2 race condition
There is code in chan_iax2.c that attempts to guarantee that only a single
active thread will handle a call number at a time. This code works once
the thread is added to an active_list of threads, but we are not currently
guaranteed that a newly activated thread will enter the active_list immediately
because it is left up to the thread to add itself after frames have been
queued to it. This means that if two frames come in for the same call number
at the same time, it is possible for them to grab two separate threads because
the first thread did not add itself to the active_list fast enough. This
causes some pretty complex problems.
This patch resolves this race condition by immediately adding an activated
thread to the active_list within the network thread and only depending on
the thread to remove itself once it is done processing the frames queued to
it. By doing this we are guaranteed that if another frame for the same call
number comes in at the same time, that this thread will immediately be found
in the active_list of threads.
Review: https://reviewboard.asterisk.org/r/720/
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After 5 years in mantis and over a year on reviewboard, SRTP support is finally
being comitted. This includes generic CHANNEL dialplan functions that work for
getting the status of whether a call has secure media or signaling as defined
by the underlying channel technology and for setting whether or not a new
channel being bridged to a calling channel should have secure signaling or
media. See doc/tex/secure-calls.tex for examples.
Original patch by mikma, updated for trunk and revised by me.
(closes issue #5413)
Reported by: mikma
Tested by: twilson, notthematrix, hemanshurpatel
Review: https://reviewboard.asterisk.org/r/191/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Very unfortunate things happen if we add an iax_frame
to the frame queue and let go of the lock before scheduling
the frame's transmit... There is a race condition that
exists where the frame can be removed from the frame_queue
and freed before the transmit is scheduled if we do not
hold on to that lock. This results in a freed frame
being scheduled for transmit later.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263151 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r250394 | dvossel | 2010-03-03 12:02:27 -0600 (Wed, 03 Mar 2010) | 16 lines
fixes problem with duplicate TXREQ packets
When Asterisk receives an IAX2 TXREQ packet, try_transfer()
will call store_by_transfercallno() to link the chan_iax2_pvt
struct into iax_transfercallno_pvts. If a duplicate TXREQ
packet is received for the same call, the pvt struct will be
linked into iax_transfercallno_pvts multiple times. This patch
fixes this. Thanks rain for debugging this and providing a patch!
(closes issue #16904)
Reported by: rain
Patches:
iax2-double-txreq-fix.diff uploaded by rain (license 327)
Tested by: rain, dvossel
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r245792 | dvossel | 2010-02-09 16:55:38 -0600 (Tue, 09 Feb 2010) | 12 lines
Fixes iaxs and iaxsl size off by one issue.
2^15 = 32768 which is the maximum allowed iax2 callnumber.
Creating the iaxs and iaxsl array of size 32768 means the maximum
callnumber is actually out of bounds. This causes a nasty crash.
(closes issue #15997)
Reported by: exarv
Patches:
iax_fix.diff uploaded by dvossel (license 671)
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When this patch was originally submitted, the code allowed for the token to be
set via a channel variable. I decided that a cleaner approach would be to
integrate it into the CHANNEL() function. Unfortunately, that is not a suitable
approach. It's not possible to get the value set on the channel soon enough
using that method. So, go back to the simple channel variable method.
(closes issue #16711)
Reported by: homesick
Patches:
iax-svn.diff uploaded by homesick (license 91)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243482 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fixes a crash on Solaris.
(closes issue #16572)
Reported by: crjw
Patches:
frame_changes.patch uploaded by crjw (license 963)
Plus several others found and fixed by me
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239074 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r238411 | dvossel | 2010-01-07 14:14:25 -0600 (Thu, 07 Jan 2010) | 10 lines
fixes crash in "scheduled_destroy" in chan_iax
A signed short was used to represent a callnumber. This is makes
it possible to attempt to access the iaxs array with a negative
index.
(closes issue #16565)
Reported by: jensvb
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r230246 | kpfleming | 2009-11-15 11:19:06 -0600 (Sun, 15 Nov 2009) | 6 lines
Correct mistaken option name in error message.
The configuration option for allowing hosts to make non-token-based calls
is 'calltokenoptional', not 'calltokenignore'. (reported on asterisk-users)
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r225243 | dvossel | 2009-10-21 15:58:08 -0500 (Wed, 21 Oct 2009) | 13 lines
IAX2: VNAK loop caused by signaling frames with no destination call number
It is possible for the PBX thread to queue up signaling frames before
a destination call number is received. This can result in signaling
frames being sent out with no destination call number. Since recent
versions of Asterisk require accurate destination callnumbers for all
Full Frames, this can cause a VNAK loop to occur. To resolve this
no signaling frames are sent until a destination callnumber is received,
and destination call numbers are now only required for iax_pvt matching
when the frame is an ACK.
Review: https://reviewboard.asterisk.org/r/413/
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r225032 | dvossel | 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009) | 20 lines
IAX/SIP shrinkcallerid option
The shrinking of caller id removes '(', ' ', ')', non-trailing '.',
and '-' from the string. This means values such as 555.5555 and
test-test result in 555555 and testtest. There are instances,
such as Skype integration, where a specific value is passed via
caller id that must be preserved unmodified. This patch makes
the shrinking of caller id optional in chan_sip and chan_iax in
order to support such cases. By default this option is on to
preserve previous expected behavior.
(closes issue #15940)
Reported by: dimas
Patches:
v2-15940.patch uploaded by dimas (license 88)
15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671)
Tested by: dvossel
Review: https://reviewboard.asterisk.org/r/408/
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r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 Oct 2009) | 20 lines
Fix ao2_iterator API to hold references to containers being iterated.
See Mantis issue for details of what prompted this change.
Additional notes:
This patch changes the ao2_iterator API in two ways: F_AO2I_DONTLOCK
has become an enum instead of a macro, with a name that fits our
naming policy; also, it is now necessary to call
ao2_iterator_destroy() on any iterator that has been
created. Currently this only releases the reference to the container
being iterated, but in the future this could also release other
resources used by the iterator, if the iterator implementation changes
to use additional resources.
(closes issue #15987)
Reported by: kpfleming
Review: https://reviewboard.asterisk.org/r/383/
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r219519 | dvossel | 2009-09-18 18:19:50 -0500 (Fri, 18 Sep 2009) | 9 lines
iax2 frame double free
The iax frame's retrans sched id was written over right
before iax2_frame_free was called. In iax2_frame_free that
retrans id is used to delete the sched item. By writing over
the retrans field before the sched item could be deleted, it was
possible for a retransmit to occur on a freed frame.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@219520 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In the course of this, I also found that the results of ast_gethostbyname
were being used incorrectly in both chan_iax2 and chan_sip, so both have
been fixed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217916 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r217806 | dvossel | 2009-09-10 16:06:07 -0500 (Thu, 10 Sep 2009) | 22 lines
IAX2 encryption regression
The IAX2 Call Token security patch inadvertently broke the use of
encryption due to the reorganization of code in the socket_process()
function. When encryption is used, an incoming full frame must first
be decrypted before the information elements can be parsed. The
security release mistakenly moved IE parsing before decryption in
order to process the new Call Token IE. To resolve this, decryption
of full frames is once again done before looking into the frame. This
involves searching for an existing callno, checking the pvt to see if
encryption is turned on, and decrypting the packet before the internal
fields of the full frame are accessed.
(closes issue #15834)
Reported by: karesmakro
Patches:
iax2_encryption_fix_1.4.diff uploaded by dvossel (license 671)
Tested by: dvossel, karesmakro
Review: https://reviewboard.asterisk.org/r/355/
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r208746 | jpeeler | 2009-07-25 01:19:50 -0500 (Sat, 25 Jul 2009) | 7 lines
Fix compiling under dev-mode with gcc 4.4.0.
Mostly trivial changes, but I did not know of any other way to fix the
"dereferencing type-punned pointer will break strict-aliasing rules" error
without creating a tmp variable in chan_skinny.
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r206385 | russell | 2009-07-14 09:48:00 -0500 (Tue, 14 Jul 2009) | 13 lines
Merged revisions 206384 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r206384 | russell | 2009-07-14 09:45:47 -0500 (Tue, 14 Jul 2009) | 6 lines
Ensure apathetic replies are sent out on the proper socket.
chan_iax2 supports multiple address bindings. The send_apathetic_reply()
function did not attempt to send its response on the same socket that the
incoming message came in on.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@206386 65c4cc65-6c06-0410-ace0-fbb531ad65f3
CEL is the new system for logging channel events. This was inspired after
facing many problems trying to represent what is possible to happen to a call
in Asterisk using CDR records. For more information on CEL, see the built in
HTML or PDF documentation generated from the files in doc/tex/.
Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard
work developing this code. Also, thanks to Matt Nicholson (mnicholson) and
Sean Bright (seanbright) for their assistance in the final push to get this
code ready for Asterisk trunk.
Review: https://reviewboard.asterisk.org/r/239/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203638 65c4cc65-6c06-0410-ace0-fbb531ad65f3
chan_sip has an option to save the sysname on rtupdate. This patch copies that same logic to chan_iax.
(closes issue #14837)
Reported by: barthpbx
Patches:
iax2-rtsavesysname.patch uploaded by barthpbx (license 744)
rt_iax.diff uploaded by dvossel (license 671)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201534 65c4cc65-6c06-0410-ace0-fbb531ad65f3
These modules all contained variables that are module-global but not system-global,
but were not marked 'static'.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200587 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Iax2 currently does not support native bridging if the timeoutms value is set. We check for that in iax2_bridge, but then set timeoutms to 0 by default. If the timeoutms is not provided it is set to -1. By setting timeoutms to 0 it is processed causing a bridging retry loop.
(closes issue #15216)
Reported by: oxymoron
Tested by: dvossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198824 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A new xml element was created to manage the AMI actions documentation,
using AstXML.
To register a manager action using XML documentation it is now possible
using ast_manager_register_xml().
The CLI command 'manager show command' can be used to show the parsed
documentation.
Example manager xml documentation:
<manager name="ami action name" language="en_US">
<synopsis>
AMI action synopsis.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(...)" /> <-- for ActionID
<parameter name="header1" required="true">
<para>Description</para>
</parameter>
...
</syntax>
<description>
<para>AMI action description</para>
</description>
<see-also>
...
</see-also>
</manager>
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196308 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes:
- CLI command handlers
- CLI command handler arguments
- AGI command handlers
- AGI command handler arguments
- Dialplan application handler arguments
- Speech engine API function arguments
In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing.
Review: https://reviewboard.asterisk.org/r/251/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r195991 | dvossel | 2009-05-21 14:04:56 -0500 (Thu, 21 May 2009) | 14 lines
Sign problem calculating timestamp for iax frame leads to no audio on the receiving peer.
There are rare cases in which a frame's delivery timestamp is slightly less than the iax2_pvt's offset. This causes the pvt's timestamp to be a small negative number, but since the timestamp value is unsigned it looks like a huge positive number. This patch checks for this negative case and sets the ms to zero. A similar check is already done right below this one in the 'else' statement.
(closes issue #15032)
Reported by: guillecabeza
Patches:
chan_iax2.c.patch_timestamp uploaded by guillecabeza (license 380)
Tested by: guillecabeza
(closes issue #14216)
Reported by: Andrey Sofronov
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r194557 | dvossel | 2009-05-14 17:59:43 -0500 (Thu, 14 May 2009) | 10 lines
IAX2 "Ghost" Channels
There is a bug tracker issue where people are reporting "Ghost" channels in their 'iax2 show channels' output. The confusion is caused by channels being listed as "(NONE)" with format "unknown". These are not channels of coarse. They are usually just pending registration or poke requests, but it is confusing output. To help make sense of this I have added two columns to 'iax2 show channels'. One shows the first message which started the transaction, and the second shows the last message sent by either side of the call. This helps diagnose why the entry exists and why it may not go away.
(closes issue #14207)
Reported by: clive18
Review: https://reviewboard.asterisk.org/r/246/
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r194685 | dvossel | 2009-05-15 10:40:37 -0500 (Fri, 15 May 2009) | 6 lines
Update to previous IAX2 "Ghost" Channels patch.
Fixed some comments made on reviewboard for the previous patch.
(issue #14207)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194833 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This scenario crept up if chan_iax2 was loaded with no configuration file present.
It would create a timer and tell it to go at an interval but the thread that normally
acknowledges it would not be created because no configuration file was present. The timer
will now be closed if no configuration file is present.
(closes issue #15014)
Reported by: madkins
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192808 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r192213 | dvossel | 2009-05-04 17:37:31 -0500 (Mon, 04 May 2009) | 11 lines
global mohinterpret setting is ignored
mohinterpret and mohsuggest global variables were not copied over during build_users and build_peers.
(closes issue #14728)
Reported by: dimas
Patches:
v1-14728.patch uploaded by dimas (license 88)
Tested by: dimas, dvossel
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192214 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r189391 | dbailey | 2009-04-20 14:10:56 -0500 (Mon, 20 Apr 2009) | 4 lines
Clean up problem with manager implementation of mmap where it was not testing against MAP_FAILED response.
Got rid of shadowed variable used in processign the mmap results.
Change test of mmap results to compare against MAP_FAILED
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@189419 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The channel drivers which have been most heavily tested with these enhancements are
chan_sip and chan_misdn. Further work is being done to add Q.SIG support and will be
introduced in a later commit. chan_skinny has code added to it here, but according
to user pj, the support on chan_skinny is not working as of now. This will be fixed in
a later commit.
A special thanks goes out to bugtracker user gareth for getting the ball rolling and
providing the initial support for this work. Without his initial work on this, this would
not have been nearly as painless as it was.
This functionality has been tested by Digium's product quality department, as well as a
customer site running thousands of calls every day. In addition, many many many many bugtracker
users have tested this, too.
(closes issue #8824)
Reported by: gareth
Review: http://reviewboard.digium.com/r/201
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This API call now waits for a special frame from the underlying channel driver to
indicate success or failure. This allows the return value to truly convey whether
the transfer worked or not. In the case of the Transfer() dialplan application this
means the value of the TRANSFERSTATUS dialplan variable is actually true.
(closes issue #12713)
Reported by: davidw
Tested by: file
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186382 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In my tests that exercised full frame handling in chan_iax2, the version with
these changes took 30% to 40% of the CPU time compared to the same test of
Asterisk trunk before these modifications.
While doing some profiling for <http://reviewboard.digium.com/r/205/>,
one function that caught my eye was network_thread() in chan_iax2.c.
After the things that I was working on there, it was the next target
for analysis and optimization. I used oprofile's source annotation
functionality and found that the loop traversing the frame queue in
network_thread() was to blame for the excessive CPU cycle consumption.
The frame_queue in chan_iax2 previously held all frames that either were
pending transmission or had been transmitted and are still pending
acknowledgment.
In network_thread(), the previous code would go back through the main
for loop after reading a single incoming frame or after being signaled
because a frame had been queued up for initial transmission. In each
iteration of the loop, it traverses the entire frame queue looking for
frames that need to be transmitted. On a busy server, this could easily
be quite a few entries.
This patch is actually quite simple. The frame_queue has become only a list
of frames pending acknowledgment. Frames that need to be transmitted are
queued up to a dedicated transmit thread via the taskprocessor API.
As a result, the code in network_thread() becomes much simpler, as its only
job is to read incoming frames.
In addition to the previously described changes, this patch includes some
additional changes to the frame_queue. Instead of one big frame_queue, now
there is a list per call number to further reduce wasted list traversals.
The biggest impact of this change is in socket_process().
For additional details on testing and test results, see the review request.
Review: http://reviewboard.digium.com/r/212/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@185432 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The ability to load/unload timing interfaces is nice, but it means that when a timer is allocated, it may come from provider A, but later provider B becomes the 'preferred' provider. If this happens, all timer API calls on the timer that was provided by provider A will actually be handed to provider B, which will say WTF and return an error.
This patch changes the timer API to include a pointer to the provider of the timer handle so that future operations on the timer will be forwarded to the proper provider.
(closes issue #14697)
Reported by: moy
Review: http://reviewboard.digium.com/r/211/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184762 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This code comes from svn/asterisk/team/russell/event_performance/.
Here is a summary of the changes that have been made, in order of both
invasiveness and performance impact, from smallest to largest.
1) Asterisk 1.6.1 introduces some additional logic to be able to handle
distributed device state. This functionality comes at a cost.
One relatively minor change in this patch is that the extra processing
required for distributed device state is now completely bypassed if
it's not needed.
2) One of the things that I noticed when profiling this code was that a
_lot_ of time was spent doing string comparisons. I changed the way
strings are represented in an event to include a hash value at the front.
So, before doing a string comparison, we do an integer comparison on the
hash.
3) Finally, the code that handles the event cache has been re-written.
I tried to do this in a such a way that it had minimal impact on the API.
I did have to change one API call, though - ast_event_queue_and_cache().
However, the way it works now is nicer, IMO. Each type of event that
can be cached (MWI, device state) has its own hash table and rules for
hashing and comparing objects. This by far made the biggest impact on
performance.
For additional details regarding this code and how it was tested, please see the
review request.
(closes issue #14738)
Reported by: russell
Review: http://reviewboard.digium.com/r/205/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184339 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The default codec configuration for chan_iax2 is bandwidth=low. I noticed
slin16 being negotiated as the codec in some test calls, but that no longer
happens after this change.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184037 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r182281 | dvossel | 2009-03-16 12:47:42 -0500 (Mon, 16 Mar 2009) | 7 lines
Randomize IAX2 encryption padding
The 16-32 byte random padding at the beginning of an encrypted IAX2 frame turns out to not be all that random at all. This patch calls ast_random to fill the padding buffer with random data. The padding is randomized at the beginning of every encrypted call and for every encrypted retransmit frame.
Review: http://reviewboard.digium.com/r/193/
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r181340 | dvossel | 2009-03-11 12:25:31 -0500 (Wed, 11 Mar 2009) | 11 lines
encrypted IAX2 during packet loss causes decryption to fail on retransmitted frames
If an iax channel is encrypted, and a retransmit frame is sent, that packet's iseqno is updated while it is encrypted. This causes the entire frame to be corrupted. When the corrupted frame is sent, the other side decrypts it and sends a VNAK back because the decrypted frame doesn't make any sense. When we get the VNAK, we look through the sent queue and send the same corrupted frame causing a loop. To fix this, encrypted frames requiring retransmission are decrypted, updated, then re-encrypted. Since key-rotation may change the key held by the pvt struct, the keys used for encryption/decryption are held within the iax_frame to guarantee they remain correct.
(closes issue #14607)
Reported by: stevenla
Tested by: dvossel
Review: http://reviewboard.digium.com/r/192/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181371 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Iax2 prune realtime had issues. If "iax2 prune realtime all" was called, it would appear like the command was successful, but in reality nothing happened. This is because the reload that was supposed to take place checks the config files, sees no changes, and does nothing. If there had been a change in the the config file, the realtime users would have been marked for deletion and everything would have been fine. Now prune_users() and prune_peers() are called instead of reload_config() to prune all users/peers that are realtime. These functions remove all users/peers with the rtfriend and delme flags set. iax2_prune_realtime() also lacked the code to properly delete a single friend. For example. if iax2 prune realtime <friend> was called, only the peer instance would be removed. The user would still remain.
(closes issue #14479)
Reported by: mousepad99
Review: http://reviewboard.digium.com/r/176/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178767 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Manager command "iaxpeers" now shows if a link is trunked and encrypted. Instead of encryption saying simply "yes" or "no", it now displays what type of encryption is enabled and if keyrotation is on or not.
(closes issue #14427)
Reported by: snuffy
Patches:
iax_show_trunks.diff uploaded by snuffy (license 35)
2009022200_iax2_show_trunkencryption.diff.txt uploaded by mvanbaak (license 7)
Tested by: mvanbaak, dvossel, snuffy
Review: http://reviewboard.digium.com/r/173/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178300 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Key rotation was enabled by default by setting the global encryption method to IAX_ENCRYPT_KEYROTATE. the problem with this is that if encryption is not enabled, and the encryption method is set to anything except 0, the peer appears to have encryption enabled when issuing a "iax2 show peers". Rather than have the key rotation bit always set by default, it is now only set when an encryption method is enabled.
(closes issue #14523)
Reported by: mvanbaak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178030 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r176354 | dvossel | 2009-02-16 17:30:52 -0600 (Mon, 16 Feb 2009) | 8 lines
Fixes issue with AST_CONTROL_SRCUPDATE not being relayed correctly during bridging
This should have been committed with rev176247, but I missed it. srcupdate frames no longer break out of the native bridge, but are not being sent to the other call leg either. This fixs that.
issue #13749
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r175597 | dvossel | 2009-02-13 14:11:55 -0600 (Fri, 13 Feb 2009) | 4 lines
Fixed iax2 key rotation backwards compatibility
Turns key rotation back on by default. Added bit into encryption IE to indicate whether or not key rotation is supported or not. If it is not supported then it is not enabled, which insures backwards compatibility. This eliminates the need for the keyrotate option in iax.conf, so it has been removed.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176248 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Turns key rotation back on by default. Added bit into encryption IE to indicate whether or not key rotation is supported or not. If it is not supported then it is not enabled, which insures backwards compatibility. This eliminates the need for the keyrotate option in iax.conf, so it has been removed.
Review: http://reviewboard.digium.com/r/159/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175597 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds forceencryption=yes as an iax.conf option. When force encryption is enabled, no unencrypted connections are allowed. This insures all connections are encrypted. This is a new feature, so CHANGES and iax.conf.sample are updated as well.
(closes issue #13285)
Reported by: sgofferj
Tested by: russell
Review: http://reviewboard.digium.com/r/150/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Key rotation breaks compatibility between (trunk/1.6.1) and (1.2/1.4/1.6.0). As a follow up to this, I am investigating possible ways to allow key rotation to be on by default and not affect the other branches, but for now it must be turned off.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is very similar to the DAHDI immediate=yes option. When the phone is picked up, instead of giving a dialtone it connects directly to the "s" extension. Changes where implemented in chan_iax2.c to directly connect to the "s" extension in the appropriate context when this option is enabled. Examples explaining its use are added to iax2.conf.sample. CHANGES has been updated as well.
(closes issue #14266)
Reported by: jcovert
Patches:
chan_iax2.c.patch-trunk uploaded by jcovert (license 551)
iax.conf.sample.patch uploaded by jcovert (license 551)
Tested by: jcovert, dvossel
Review: http://reviewboard.digium.com/r/143/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174046 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit expands the Asterisk scheduler API to include a common implementation
of a scheduler context being processed by a dedicated thread. chan_iax2 has been
updated to use this new code. Also, as a result, this resolves some race
conditions related to the previous chan_iax2 scheduler handling.
Related to rev 171452 which resolved the same issues in 1.4.
Code from team/russell/sched_thread2
Review: http://reviewboard.digium.com/r/129/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fixes issue with IAX2 transfers not taking place. As it was, a call that was being transfered would never be handed off correctly to the call ends because of how call numbers were stored in a hash table. The hash table, "iax_peercallno_pvt", storing all the current call numbers did not take into account the complications associated with transferring a call, so a separate hash table was required. This second hash table "iax_transfercallno_pvt" handles calls being transfered, once the call transfer is complete the call is removed from the transfer hash table and added to the peer hash table resuming normal operations. Addition functions were created to handle storing, removing, and comparing items in the iax_transfercallno_pvt table. The changes reverted in 116884 caused backwards compatibility issues involving iax2 transfer with 1.6.0, 1.4, and 1.2.
(closes issue #13468)
Reported by: nicox
Tested by: dvossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173502 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r160003 | russell | 2008-12-01 11:27:30 -0600 (Mon, 01 Dec 2008) | 6 lines
Apply some logic used in iax2_indicate() to iax2_setoption(), as well, since they
both have the potential to send control frames in the middle of call setup. We
have to wait until we have received a message back from the remote end before
we try to send any more frames. Otherwise, the remote end will consider it
invalid, and we'll get stuck in an INVAL/VNAK storm.
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r159808 | kpfleming | 2008-11-29 10:58:29 -0600 (Sat, 29 Nov 2008) | 7 lines
update dev-mode compiler flags to match the ones used by default on Ubuntu Intrepid, so all developers will see the same warnings and errors
since this branch already had some printf format attributes, enable checking for them and tag functions that didn't have them
format attributes in a consistent way
------------------------------------------------------------------------
in addition:
move some format attributes from main/utils.c to the header files they belong in, and fix up references to the relevant functions based on new compiler warnings
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159818 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r159316 | murf | 2008-11-25 15:41:10 -0700 (Tue, 25 Nov 2008) | 15 lines
(closes issue #12694)
Reported by: yraber
Patches:
12694.2nd.diff uploaded by murf (license 17)
Tested by: murf, laurav
Thanks to file (Joshua Colp) for his IAX fix.
the change to cdr.c allows no-answer to percolate
up into CDR's, and feels like the right place to
locate this fix; if BUSY is done here, no-answer
should be, too.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159360 65c4cc65-6c06-0410-ace0-fbb531ad65f3
it would be best to maintain API compatibility. Instead, this commit introduces
ao2_callback_data() which is functionally identical to ao2_callback() except
that it allows you to pass arbitrary data to the callback.
Reviewed by Mark Michelson via ReviewBoard:
http://reviewboard.digium.com/r/64
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r156229 | tilghman | 2008-11-12 12:39:21 -0600 (Wed, 12 Nov 2008) | 11 lines
Revert revision 132506, since it occasionally caused IAX2 HANGUP packets not
to be sent, and instead, schedule a task to destroy the iax2 pvt structure
10 seconds later. This allows the IAX2 HANGUP packet to be queued,
transmitted, and ACKed before the pvt is destroyed.
(closes issue #13645)
Reported by: dzajro
Patches:
20081111__bug13645__3.diff.txt uploaded by Corydon76 (license 14)
Tested by: vazir
Reviewed: http://reviewboard.digium.com/r/51/
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ao2_callback and ao2_find). Currently, passing OBJ_POINTER to either
of these mandates that the passed 'arg' is a hashable object, making
searching for an ao2 object based on outside criteria difficult.
Reviewed by Russell and Mark M. via ReviewBoard:
http://reviewboard.digium.com/r/36/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@155401 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r154365 | tilghman | 2008-11-04 14:49:33 -0600 (Tue, 04 Nov 2008) | 9 lines
On busy systems, it's possible for the values checked within a single line
of code to change, unless the structure is locked to ensure a consistent
state.
(closes issue #13717)
Reported by: kowalma
Patches:
20081102__bug13717.diff.txt uploaded by Corydon76 (license 14)
Tested by: kowalma
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154366 65c4cc65-6c06-0410-ace0-fbb531ad65f3
for more details of this command.
(closes issue #13326)
Reported by: ib2
Patches:
bug13326_trunk_20080822.diff uploaded by snuffy (license 35)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: nickpeirson
Patches:
pbx.c.patch uploaded by nickpeirson (license 579)
replace_bzero+bcopy.patch uploaded by nickpeirson (license 579)
Tested by: nickpeirson, murf
1. replaced all refs to bzero and bcopy to memset and memmove instead.
2. added a note to the CODING-GUIDELINES
3. add two macros to asterisk.h to prevent bzero, bcopy from creeping
back into the source
4. removed bzero from configure, configure.ac, autoconfig.h.in
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147807 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r146799 | tilghman | 2008-10-06 15:52:04 -0500 (Mon, 06 Oct 2008) | 8 lines
Dialplan functions should not actually return 0, unless they have modified the
workspace. To signal an error (and no change to the workspace), -1 should be
returned instead.
(closes issue #13340)
Reported by: kryptolus
Patches:
20080827__bug13340__2.diff.txt uploaded by Corydon76 (license 14)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
when a file is invalid from when a file is missing. This is most important when
we have two configuration files. Consider the following example:
Old system:
sip.conf users.conf Old result New result
======== ========== ========== ==========
Missing Missing SIP doesn't load SIP doesn't load
Missing OK SIP doesn't load SIP doesn't load
Missing Invalid SIP doesn't load SIP doesn't load
OK Missing SIP loads SIP loads
OK OK SIP loads SIP loads
OK Invalid SIP loads incompletely SIP doesn't load
Invalid Missing SIP doesn't load SIP doesn't load
Invalid OK SIP doesn't load SIP doesn't load
Invalid Invalid SIP doesn't load SIP doesn't load
So in the case when users.conf doesn't load because there's a typo that
disrupts the syntax, we may only partially load users, instead of failing with
an error, which may cause some calls not to get processed. Worse yet, the old
system would do this with no indication that anything was even wrong.
(closes issue #10690)
Reported by: dtyoo
Patches:
20080716__bug10690.diff.txt uploaded by Corydon76 (license 14)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142992 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Changes applied by this patch:
- Fix autocompletion in 'sip prune realtime', sip peers where never auto completed. Now we complete this command with:
'sip prune realtime peer' -> all | like | sip peers
Also I have modified the syntax in the usage, was wrong...
- Pass ast_cli_args->argv and ast_cli_args->argc while running autocompletion on CLI commands (CLI_GENERATE).
With this we avoid comparisons on ast_cli_args->line like this:
strcasestr(a->line, " description")
strcasestr(a->line, "descriptions ")
strcasestr(a->line, "realtime peer"), and so on..
Making the code more confusing (check the spaces in description!).
The only thing we must be sure is to first check a->pos or a->argc.
- Fix 'iax2 prune realtime' autocompletion, now we autocomplete this command with 'all' & 'iax2 peers', check a look that iax2 peers where all the peers, now only the ones in the cache..
(closes issue #13133)
Reported by: eliel
Patches:
clichanges.patch uploaded by eliel (license 64)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@141464 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r140488 | mmichelson | 2008-08-29 12:34:17 -0500 (Fri, 29 Aug 2008) | 22 lines
After working on the ao2_containers branch, I noticed
something a bit strange. In all cases where we provide
a callback function to ao2_container_alloc, the callback
function would only return 0 or CMP_MATCH. After inspecting
the ao2_callback() code carefully, I found that if you're
only looking for one specific item, then you should return
CMP_MATCH | CMP_STOP. Otherwise, astobj2 will continue
traversing the current bucket until the end searching for
more matches.
In cases like chan_iax2 where in 1.4, all the peers are
shoved into a single bucket, this makes for potentially
terrible performance since the entire bucket will be
traversed even if the peer is one of the first ones come
across in the bucket.
All the changes I have made were for cases where the
callback function defined was passed to ao2_container_alloc
so that calls to ao2_find could find a unique instance
of whatever object was being stored in the container.
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r140051 | russell | 2008-08-26 10:27:23 -0500 (Tue, 26 Aug 2008) | 15 lines
Fix a race condition with the IAX scheduler thread. A lock and condition are
used here to allow newly scheduled tasks to wake up the scheduler just in case
the new task needs to run sooner than the current wakeup time when the thread
is sleeping. However, there was a race condition such that a newly scheduled
task would not properly wake up the scheduler or affect the wake up period.
The order of execution would have been:
1) Scheduler thread determines wake up time of N ms.
2) Another thread schedules a task and signals the condition, with an
execution time of < N ms.
3) Scheduler thread locks and goes to sleep for N ms.
By moving the sleep time determination to inside the critical section, this
possibility is avoided.
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r135747 | tilghman | 2008-08-05 16:34:46 -0500 (Tue, 05 Aug 2008) | 9 lines
In a conversion to use ast_strlen_zero, the meaning of the flag IAX_HASCALLERID
was perverted. This change reverts IAX2 to the original meaning, which was,
that the callerid set on the client should be overridden on the server, even if
that means the resulting callerid is blank. In other words, if you set
"callerid=" in the IAX config, then the callerid should be overridden to blank,
even if set on the client. Note that there's a distinction, even on realtime,
between the field not existing (NULL in databases) and the field existing, but
set to blank (override callerid to blank).
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135748 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This set of changes enhances IAX2 encryption support by adding key rotation
to provide enhanced security. The key used for encryption is rotated right
after the call gets set up, and then again every few minutes. This was
discussed at the last AstriDevCon. For interoperability with older versions
of Asterisk, there is an option that disables key rotation.
(closes issue #13018)
Reported by: bbryant
Patches:
07072008__iax2_key_rotation.diff uploaded by bbryant (license 36)
Tested by: russell, bbryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
commands that conflicted with adding new features to the newer debug
commaands.
(closes issue #13103)
Reported by: mvanbaak
Patches:
2008071901__issue13103_iax2_set_debug_peer.diff uploaded by
mvanbaak (license 7)
Tested by: bbryant, mvanbaak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@132470 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r130889 | tilghman | 2008-07-14 18:59:13 -0500 (Mon, 14 Jul 2008) | 8 lines
Override the callerid in all cases when the callerid is set in the user, not
just when a remote callerid is set. Also, if not set in the user, allow the
remote CallerID to pass through.
(closes issue #12875)
Reported by: dimas
Patches:
20080714__bug12875.diff.txt uploaded by Corydon76 (license 14)
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r130169 | tilghman | 2008-07-11 13:51:56 -0500 (Fri, 11 Jul 2008) | 7 lines
Ensure that a destination callno of 0 will not match for frames that do not
start a dialog (new, lagrq, and ping).
(closes issue #12963)
Reported by: russellb
Patches:
chan_iax2_dup_new_fix4.patch uploaded by jpgrayson (license 492)
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r128795 | russell | 2008-07-07 17:41:48 -0500 (Mon, 07 Jul 2008) | 8 lines
Fix handling of when a pvt disappears. Properly return the pvt locked
and don't hold the pvt lock while destroying the ast_channel.
(closes issue #13014)
Reported by: jpgrayson
Patches:
chan_iax2_ast_iax2_new2.patch uploaded by jpgrayson (license 492)
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r128639 | mmichelson | 2008-07-07 12:02:28 -0500 (Mon, 07 Jul 2008) | 10 lines
By using the iaxdynamicthreadcount to identify a thread, it was possible
for thread identifiers to be duplicated. By using a globally-unique monotonically-
increasing integer, this is now avoided.
(closes issue #13009)
Reported by: jpgrayson
Patches:
chan_iax2_dyn_threadnum.patch uploaded by jpgrayson (license 492)
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They want (char *)NULL as sentinel.
An example is OpenBSD (confirmed on 4.3) that ships with gcc 3.3.4
This commit introduces a contstant SENTINEL which is declared as:
#define SENTINEL ((char *)NULL)
All places I could test compile on my openbsd system are converted.
Update CODING-GUIDELINES to tell about this constant.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@124127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
function and begin the transition from SIPCHANINFO() to just using CHANNEL().
(closes issue #12856)
Reported by: mostyn
Patches:
iax_and_sip_channel_info.patch uploaded by mostyn (license 398)
(with some additional cleanup by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Convert chan_iax2 to use the timing API
- Convert usage of timing in the core to use the timing API instead of
using DAHDI directly
- Make a change to the timing API to add the set_rate() function
- change the timing core to use a rwlock
- merge a timing implementation, res_timing_dahdi
Basic testing was successful using res_timing_dahdi
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122523 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Richard Mudgett
David Vossel
Russell Bryant
Leif Madsen
Tilghman Lesher
Paul Belanger
Jason Parker
Jeff Peeler
Brett Bryant
Terry Wilson
Mark Michelson
Eliel C. Sardanons
Tzafrir Cohen
Kevin P. Fleming
Joshua Colp
Doug Bailey
Steve Murphy
Sean Bright
Michiel van Baak