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r350788 | kpfleming | 2012-01-14 09:22:33 -0600 (Sat, 14 Jan 2012) | 8 lines
Ensure that two prerequisites are properly installed on Debian-style distributions.
* Don't specify a specific version of libgmime; newer versions are available
now and acceptable.
* Install libsrtp so that res_srtp can be built.
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r350789 | kpfleming | 2012-01-14 09:23:32 -0600 (Sat, 14 Jan 2012) | 3 lines
Correct some 'set-but-not-used' variable warnings.
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There are many benefits to making the ast_channel an opaque handle, from
increasing maintainability to presenting ways to kill masquerades. This patch
kicks things off by taking things a field at a time, renaming the field to
'__do_not_use_${fieldname}' and then writing setters/getters and converting the
existing code to using them. When all fields are done, we can move ast_channel
to a C file from channel.h and lop off the '__do_not_use_'.
This patch sets up main/channel_interal_api.c to be the only file that actually
accesses the ast_channel's fields directly. The intent would be for any API
functions in channel.c to use the accessor functions. No more monkeying around
with channel internals. We should use our own APIs.
The interesting changes in this patch are the addition of
channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to
channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to
use accessor functions when ast_channel is really opaque), and some re-working
of the way channel iterators/callbacks are handled so as to avoid creating fake
ast_channels on the stack to pass in matching data by directly accessing fields
(since "name" is a stringfield and the fake channel doesn't init the
stringfields, you can't use the ast_channel_name_set() function). I went with
ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a
setter.
The majority of the grunt-work for this change was done by writing a semantic
patch using Coccinelle ( http://coccinelle.lip6.fr/ ).
Review: https://reviewboard.asterisk.org/r/1655/
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r346763 | may | 2011-12-02 20:42:32 +0400 (Fri, 02 Dec 2011) | 14 lines
Merged revisions 346762 via svnmerge from
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r346762 | may | 2011-12-02 20:19:19 +0400 (Fri, 02 Dec 2011) | 7 lines
process null frame pointer returned by ast_rtp_instance_read correctly
(closes issue ASTERISK-16697)
Reported by: under
Patches:
segfault.diff (License #5871) patch uploaded by under
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Merged revisions 346239 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r337487 | irroot | 2011-09-22 11:26:26 +0200 (Thu, 22 Sep 2011) | 16 lines
Merged revisions 337486 via svnmerge from
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r337486 | irroot | 2011-09-22 11:22:26 +0200 (Thu, 22 Sep 2011) | 10 lines
If IP address is used in chan_h323 host parameter of peer configuration.
module tries to resolve IP address to IP address and fails.
Simple fix to set family of socket this is a hangover from ipv6 changes.
(closes issue ASTERISK-18237)
(issue ASTERISK-17278)
(issue ASTERISK-17500)
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r336500 | irroot | 2011-09-19 15:31:50 +0200 (Mon, 19 Sep 2011) | 19 lines
Merged revisions 336499 via svnmerge from
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r336499 | irroot | 2011-09-19 15:27:52 +0200 (Mon, 19 Sep 2011) | 13 lines
A long time ago in a galaxy far far away a IPv6 update was made,
chan_h323 was not updated causeing all to flee to chan_ooh323.
the brave Jedi [asterisk developers] pondered this miscarrige of justice
and restored order to the force for the sake of closing out 2 old issues.
(closes issue ASTERISK-17278)
(closes issue ASTERISK-17500)
Reported by: dread, sybasesql
Tested by: irroot
Reviewed by: IRC (russellb, kpfleming)
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r335078 | mjordan | 2011-09-09 11:27:01 -0500 (Fri, 09 Sep 2011) | 29 lines
Merged revisions 335064 via svnmerge from
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r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) | 23 lines
Updated SIP 484 handling; added Incomplete control frame
When a SIP phone uses the dial application and receives a 484 Address
Incomplete response, if overlapped dialing is enabled for SIP, then
the 484 Address Incomplete is forwarded back to the SIP phone and the
HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete
application dialplan logic was automatically triggered; now, explicit
dialplan usage of the application is required.
Additionally, this patch adds a new AST_CONTOL_FRAME type called
AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame,
it is an indication that the dialplan expects more digits back from the
device. If the device supports overlap dialing it should attempt to
notify the device that the dialplan is waiting for more digits; otherwise,
it can handle the frame in a manner appropriate to the channel driver.
(closes issue ASTERISK-17288)
Reported by: Mikael Carlsson
Tested by: Matthew Jordan
Review: https://reviewboard.asterisk.org/r/1416/
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r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines
Merged revisions 328209 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
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r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines
Introduce <support_level> tags in MODULEINFO.
This change introduces MODULEINFO into many modules in Asterisk in order to show
the community support level for those modules. This is used by changes committed
to menuselect by Russell Bryant recently (r917 in menuselect). More information about
the support level types and what they mean is available on the wiki at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
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r317478 | russell | 2011-05-05 17:53:45 -0500 (Thu, 05 May 2011) | 12 lines
Fix some consistency issues with jitterbuffer config.
Store the defaults noted in the sample config files in the jitterbuffer config
data structure. This makes the CLI commands that output these settings show
the right thing. Also only show the settings that are relevant in the settings
CLI commands, based on which jitterbuffer is selected and whether it's enabled.
(closes issue #19083)
Reported by: rgagnon
Patches:
issue-19083-trunk-r313139.diff uploaded by rgagnon (license 1202)
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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/
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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/
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r291758 | pabelanger | 2010-10-14 11:15:12 -0400 (Thu, 14 Oct 2010) | 11 lines
Add the ability for ast_find_ourip to return IPv4, IPv6 or both.
While testing chan_gtalk I noticed jabber was using my IPv6 address
and not IPv4. When using bindaddr=0.0.0.0 it is possible for ast_find_ourip()
to return both IPv6 and IPv4 results. Adding a family parameter gives you
the ablility to choose.
Since jabber/gtalk/h323 do not support IPv6, we should only return IPv4 results.
Review: https://reviewboard.asterisk.org/r/973/
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Expand the ani field in ast_party_caller and ast_party_connected_line to
an ast_party_id.
This is an extension to the ast_callerid restructuring patch in review:
https://reviewboard.asterisk.org/r/702/
Review: https://reviewboard.asterisk.org/r/744/
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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/
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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
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r273793 | tilghman | 2010-07-02 16:36:39 -0500 (Fri, 02 Jul 2010) | 9 lines
Have the DEADLOCK_AVOIDANCE macro warn when an unlock fails, to help catch potentially large software bugs.
(closes issue #17407)
Reported by: pdf
Patches:
20100527__issue17407.diff.txt uploaded by tilghman (license 14)
Review: https://reviewboard.asterisk.org/r/751/
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(Also fix the typos in the comments)
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SWP-1229
ABE-2161
* Ensure chan_local.c:local_call() will not leak cid.cid_dnid when
copying.
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r255409 | russell | 2010-03-30 15:56:00 -0500 (Tue, 30 Mar 2010) | 2 lines
Don't kill Asterisk if the H323 listener does not start.
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This change basically reverts the change reviewed in
https://reviewboard.asterisk.org/r/374/ and instead limits the
updating of the RTP synchronization source to only those times when we
detect that the other side of the conversation has changed the ssrc.
The problem is that SRCUPDATE control frames are sent many times where
we don't want a new ssrc, including whenever Asterisk has to send DTMF
in a normal bridge. This is also not the first time that this mistake
has been made. The initial implementation of the ast_rtp_new_source
function also changed the ssrc--and then it was removed because of
this same issue. Then, we put it back in again to fix a different
issue. This patch attempts to only change the ssrc when we see that
the other side of the conversation has changed the ssrc.
It also renames some functions to make their purpose more clear.
Review: https://reviewboard.asterisk.org/r/540/
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When configuring the adaptive jitterbuffer, the target_extra
value not only could not be set from the configuration, but was
not even being set to its proper default. This value is required
in order for the adaptive jitterbuffer to work correctly. To resolve
this a config option has been added to expose this value to the conf
files, and a default value is provided when no config specific value
is present.
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In the process of swapping ULAW to a place in the extended codec space, we
found several unhandled cases, where a 32-bit integer was still being used to
handle a codec field. Most of these have been fixed with this commit, although
there is at least one case (codec_dahdi) which depends upon outside headers to
be altered before a conversion can be made.
(Fixes AST-278, SWP-459)
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Reviewboard: https://reviewboard.asterisk.org/r/416/
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This fixes a crash reported in #asterisk-dev where chan_mgcp unexpectedly
allocated an RTP instance from res_rtp_multicast, since by not specifying an
engine, you get the first one in the list of engines.
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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/
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These modules all contained variables that are module-global but not system-global,
but were not marked 'static'.
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Previously, packetization settings were ignored and now they are not. A new
config option 'autoframing' has been added to mirror the way chan_sip handles
it. Turning on the autoframing option (available both as a global option or per
peer) overrides the local settings with the remote packetization settings.
Testing was performed with varying packetization levels with the following
codecs: ulaw, alaw, gsm, and g729.
Also, an unrelated config reload issue has been fixed in the case of the config
file not changing.
(closes issue #12415)
Reported by: pj
Patches:
2009012200_h323packetization.diff.txt uploaded by mvanbaak (license 7),
modified by me
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Asterisk.
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
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This API provides a generic way for multiple RTP stacks to be
integrated into Asterisk. Right now there is only one present, res_rtp_asterisk,
which is the existing Asterisk RTP stack. Functionality wise this commit
performs the same as previously. API documentation can be viewed in the
rtp_engine.h header file.
Review: http://reviewboard.digium.com/r/209/
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Chan_h323 can now be compiled against both the previously supported versions of
OpenH323 as well as the current H.323 Plus (version 1.20.2). The configure
script has been modified to look in the default install location of h323 to
hopefully help avoid using the environment variables OPENH323DIR and PWLIBDIR.
Also, the CLI command "h323 show version" has been added which indicates which
version of h323 is in use.
(closes issue #11261)
Reported by: vhatz
Patches:
asterisk-1.6.0.6-h323plus.patch uploaded by jthurman (license 614)
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G.722.1C (also known as Siren7 and Siren14)
This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported.
Along the way, some related work was done:
1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way.
2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec.
3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result).
Review: http://reviewboard.digium.com/r/158/
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r154263 | tilghman | 2008-11-04 12:58:05 -0600 (Tue, 04 Nov 2008) | 3 lines
Make the monitor thread non-detached, so it can be joined (suggested by Russell
on -dev list).
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branch, and add the ones needed for all the new code here too
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r152958 | tilghman | 2008-10-30 15:33:28 -0500 (Thu, 30 Oct 2008) | 3 lines
Cannot join detached threads. See http://www.opengroup.org/onlinepubs/000095399/functions/pthread_join.html
(Closes issue #13400)
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Thanks rmudget for letting me know and providing hints on how to fix it best.
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This work is done by lmadsen, junky and mvanbaak
during AstriDevCon.
This is the second audit the CLI got, and
this time lmadsen made sure he had _ALL_ modules
loaded that have CLI commands in them.
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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)
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of the channel lock
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r123113 | tilghman | 2008-06-16 14:50:12 -0500 (Mon, 16 Jun 2008) | 2 lines
Port "hasvoicemail" change from SIP to other channel drivers
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- make data member of the ast_frame struct a named union instead of a void
Recently the ast_queue_hangup function got a new parameter, the hangupcause
Feedback came in that this is no good and that instead a new function should be created.
This I did.
The hangupcause was stored in the seqno member of the ast_frame struct. This is not very
elegant, and since there's already a data member that one should be used.
Problem is, this member was a void *.
Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone
wants to store another type in there in the future.
This commit is so massive, because all ast_frame.data uses have to be
altered to ast_frame.data.data
Thanks russellb and kpfleming for the feedback.
(closes issue #12674)
Reported by: mvanbaak
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(closes issue #11328)
Reported by: rain
Patches:
20071207__pass_cause_in_hangup_control_frame.diff.txt uploaded by Corydon76 (license 14)
brought up-to-date to trunk by me
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r114120 | qwell | 2008-04-14 13:31:57 -0500 (Mon, 14 Apr 2008) | 7 lines
The call_token on the pvt can occasionally be NULL, causing a crash.
If it is NULL, we can skip this channel, since it can't the one we're looking for.
(closes issue #9299)
Reported by: vazir
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(closes issue #11901)
Reported by: pj
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