Age | Commit message (Collapse) | Author | Files | Lines |
|
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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249893 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Reviewboard: https://reviewboard.asterisk.org/r/416/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@227580 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
JABBER_RECEIVE (along with JabberSend) makes Asterisk interact with users over
XMPP to process calls.
SendText can be used instead of JabberSend in the context of XMPP based voice
channels (chan_gtalk and chan_jingle).
(closes issue #12569)
Reported by: eech55
Tested by: phsultan, asannucci, lmadsen, jtodd, maxgo
Review: https://reviewboard.asterisk.org/r/88/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@220457 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@211732 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
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: http://svn.digium.com/svn/asterisk/trunk@203638 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187772 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
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/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@186078 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
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: http://svn.digium.com/svn/asterisk/trunk@142992 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
as RSW since i am too lazy to keep typing it all out). This time a few of
the channels.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@136888 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(closes issue #13054)
Reported by: pabelanger
Patches:
ARRAY_LEN.patch2 uploaded by pabelanger (license 224)
Tested by: seanbright
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@130129 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
jabber.conf). The actual connection is made when a call comes in
Asterisk.
Apply this fix to Jingle too.
Fix the ast_aji_get_client function that was not able to retrieve an
XMPP client from its JID.
(closes issue #12085)
Reported by: junky
Tested by: phsultan
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@119741 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Jingle is completely standardized, we can set those namespaces to their
final values.
Added two attributes to the jingle_pvt struct to store the content
name attributes. Reported by Robert McQueen on Telepathy's framework
mailing list :
http://lists.freedesktop.org/archives/telepathy/2008-May/001971.html
Keeping working on our Jingle stack!
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118644 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118614 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
- 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
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117802 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r114708 | tilghman | 2008-04-27 23:47:39 -0500 (Sun, 27 Apr 2008) | 5 lines
When modules are embedded, they take on a different name, without the ".so"
extension. Specifically check for this name, when we're checking if a module
is loaded.
(Closes issue #12534)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114709 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(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
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114637 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Reported by: oej
Tested by: jpeeler
This patch implements multiple parking lots for parked calls. The default parkinglot is used by default, however setting the channel variable PARKINGLOT in the dialplan will allow use of any other configured parkinglot. See configs/features.conf.sample for more details on setting up another non-default parkinglot. Also, one can (currently) set the default parkinglot to use in the driver configuration file via the parkinglot option.
Patch initially written by oej, brought up to date and finalized by mvanbaak, and then stabilized and converted to astobj2 by me.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114487 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r107714 | qwell | 2008-03-11 15:49:56 -0500 (Tue, 11 Mar 2008) | 5 lines
Copy voicemail dependency logic for res_adsi to chan_gtalk and chan_jingle (for jabber).
(closes issue #12014)
Reported by: junky
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@107718 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
done inside iks_delete), thus making the code conform with coding guidelines.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105263 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89512 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
were included almost everywhere.
Remove some of the instances.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89424 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89349 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
build times - tested, there is no measureable difference before and
after this commit.
In this change:
use asterisk/compat.h to include a small set of system headers:
inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h,
stdlib.h, alloca.h, stdio.h
Where available, the inclusion is conditional on HAVE_FOO_H as determined
by autoconf.
Normally, source files should not include any of the above system headers,
and instead use either "asterisk.h" or "asterisk/compat.h" which does it
better.
For the time being I have left alone second-level directories
(main/db1-ast, etc.).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89333 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
- Remove the AST_FORMAT_MAX_* types, as these are consuming 3 out of our available 32 bits.
- Add a native slin16 type, so that 16kHz codecs can translate without losing resolution.
(This doesn't affect anything immediately, until another codec has wb support.)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89071 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Closes issue #9972
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89041 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88184 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #11130)
(closes issue #11132)
........
r87906 | qwell | 2007-10-31 16:16:20 -0500 (Wed, 31 Oct 2007) | 4 lines
Don't try to allocate memory that we're just going to re-allocate later anyways.
Issues 11130 and 11132, patch by eliel.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@87907 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
didn't make much sense
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86820 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Closes issue #11039, as suggested by seanbright.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86536 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85787 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
- jingle reload ;
- jingle show channels.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85778 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
SIP Phone 1 --- [chan_sip]Asterisk 1[chan_jingle] --- [chan_jingle]Asterisk 2[chan_sip] --- SIP Phone 2
Modifications :
- set bridge type to partial ;
- process media candidates from the remote peer properly.
Now we have Jingle audio, at least between two Asterisk Jingle
clients.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85777 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85555 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85140 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
No real Jingle implementation being available, testing was made using
two Asterisk servers relaying SIP calls over their Jingle channels:
SIP Phone 1 --- [chan_sip]Asterisk 1[chan_jingle] --- [chan_jingle]Asterisk 2[chan_sip] --- SIP Phone 2
Thus, it was possible to test the code in both ways, and make the
Jingle channel comply with the latest specifications. No sound available yet.
Main modifications include :
- modified the 'jingle_candidate' structure and the
'jingle_create_candidates' function according to XEP-0176 ;
- modified the 'jingle_action' function in order to properly terminate
a Jingle session, in conformance with XEP-0166 ;
- modified username format used in STUN requests ;
- actually make the bindaddr configuration field useable.
Todo :
- set audio paths up (no native bridging) ;
- make the CLI gtalk functions available to jingle ;
- clean up the storage space used in strings.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@83743 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
is handled by chan_gtalk.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@83076 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@83072 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
specifications), XEP-0167 (Jingle Audio via RTP) and XEP-0176 (Jingle ICE
Transport).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@83055 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@82373 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@82320 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
As both specifications are in the Experimental status, the namespaces
specified therein shall be of the form
"http://www.xmpp.org/extensions/xep-XXXX.html#ns".
See the Namespace issuance section in XEP-0053 :
http://www.xmpp.org/extensions/xep-0053.html#namespaces
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@82314 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@82312 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@79747 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r79174 | file | 2007-08-13 11:18:04 -0300 (Mon, 13 Aug 2007) | 4 lines
(closes issue #10437)
Reported by: haklin
Don't set the callerid name and number a second time on a newly created channel. ast_channel_alloc itself already sets it and setting it twice would cause a memory leak.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@79175 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
scalability and is done in such a way that we should be able to add support for other poll() replacements.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@78683 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@72358 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r70084 | russell | 2007-06-19 14:13:45 -0500 (Tue, 19 Jun 2007) | 3 lines
Only attempt to queue a hangup on the owner channel if it actually exists.
(issue #9795, patch from zandbelt)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@70088 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(issue #9957, patches from mvanbaak, caio1982, critch, and dimas)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@69327 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@66981 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@66175 f38db490-d61c-443f-a65b-d21fe96a405b
|