Age | Commit message (Collapse) | Author |
|
When an inbound SDP offer is received, Asterisk currently makes a few
incorrection assumptions:
(1) If the offer contains more than a single audio/video stream, Asterisk will
reject the entire stream with a 488. This is an overly strict response;
generally, Asterisk should accept the media streams that it can accept and
decline the others.
(2) If the offer contains a declined media stream, Asterisk will attempt to
process it anyway. This can result in attempting to match format
capabilities on a declined media stream, leading to a 488. Asterisk should
simply ignore declined media streams.
(3) Asterisk will currently attempt to handle offers with AVPF with
use_avpf=No/AVP with use_avpf=Yes. This mismatch results in invalid SDP
answers being sent in response. If there is a mismatch between the media
type being offered and the configuration, Asterisk must reject the offer
with a 488.
This patch does the following:
* Asterisk will accept SDP offers with at least one media stream that it can
use. Some WARNING messages have been dropped to NOTICEs as a result.
* Asterisk will not accept an offer with a media type that doesn't match its
configuration.
* Asterisk will ignore declined media streams properly.
#SIPit31
Review: https://reviewboard.asterisk.org/r/4063/
ASTERISK-24122 #close
Reported by: James Van Vleet
ASTERISK-24381 #close
Reported by: Matt Jordan
........
Merged revisions 425868 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425879 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This enforces that res_pjsip, res_pjsip_session, and res_pjsip_pubsub
have loaded properly before attempting to load any modules that depend
on them since the module loader system is not currently capable of
resolving module dependencies on its own.
ASTERISK-24312 #close
Reported by: Dafi Ni
Review: https://reviewboard.asterisk.org/r/4062/
........
Merged revisions 425690 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425691 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Performing a directed call pickup resulted in a deadlock when PJSIP
channels were involved.
A masquerade needs to hold onto the channel locks while it swaps channel
information between the two channels involved in the masquerade. With
PJSIP channels, the fixup routine needed to push a fixup task onto the
PJSIP channel's serializer. Unfortunately, if the serializer was also
processing a task that needed to lock the channel, you get deadlock.
* Added a new control frame that is used to notify the channels that a
masquerade is about to start and when it has completed.
* Added the ability to query taskprocessors if the current thread is the
taskprocessor thread.
* Added the ability to suspend/unsuspend the PJSIP serializer thread so a
masquerade could fixup the PJSIP channel without using the serializer.
ASTERISK-24356 #close
Reported by: rmudgett
Review: https://reviewboard.asterisk.org/r/4034/
........
Merged revisions 424471 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424472 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Due to the architecture of how media streams are handled each individual
handler adds connection details (IP address) for it. The first media stream
is then used as the top level SDP connection line. In practice each
line ends up being the same so to reduce the SDP size stream-level connection
information is also added to the SDP if it differs from the top level SDP
connection line.
........
Merged revisions 424414 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424415 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Replace code using ast_uuid_generate() with simpler and faster code using
ast_uuid_generate_str(). The new code avoids a malloc(), free(), and
copy.
........
Merged revisions 424103 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424105 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
There are certain situations which no checks existed for which need to prevent
session refreshes. This includes sending a session refresh with SDP before SDP
negotiation has completed and sending a session refresh before the dialog itself
has been established. Checks for these have been added.
Additionally COLP related UPDATEs were including SDP when it is not needed.
Review: https://reviewboard.asterisk.org/r/4008/
........
Merged revisions 424056 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424057 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
Merged revisions 423172 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423173 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
media has been negotiated.
Testsuite tests will occasionally fail because on reception of a 200 OK SIP response,
an AST_CONTROL_ANSWER frame is queued prior to when media has finished being
negotiated. This is because session supplements are called into before PJSIP's
inv_session code has told us that media has been updated. Sometimes the queued answer
frame is handled by the PBX thread before the ensuing media negotiations occur, causing
a test failure.
As it turns out, there is another place that session supplements could be called into, which is
after media has finished getting negotiated. What this commit introduces is a means for session
supplements to indicate when they wish to be called into when handling an incoming SIP response.
By default, all session supplements will be run at the same point that they were prior to this
commit. However, session supplements may indicate that they wish to be handled earlier than
normal on redirects, or they may indicate they wish to be handled after media has been negotiated.
In this changeset, two session supplements have been updated to indicate a preference for when
they should be run: res_pjsip_diversion executes before handling redirection in order to get
information from the Diversion header, and chan_pjsip now handles responses to INVITEs after
media negotiation to fix the race condition mentioned previously.
ASTERISK-24212 #close
Reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/3930
........
Merged revisions 422536 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422542 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Using the hostname in the SDP origin line may not satisfy the requirement
of RFC 4566 that we use a FQDN or IP address. This change has us use the
same information from the SDP connection line if possible. If not possible,
we'll use the configured media address. And if that's not possible, we use
the result of a PJLIB call to get the IP address of ourself.
ASTERISK-23994 #close
Reported by Private Name
Review: https://reviewboard.asterisk.org/r/3925
........
Merged revisions 421796 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421797 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
A calls B
B answers
B SIP attended transfers to C
C answers, B and C can see each other's connected line information
B completes the transfer
A has number but no name connected line information about C
while C has the full information about A
I examined the incoming and outgoing party id information handling of
chan_pjsip and found several issues:
* Fixed ast_sip_session_create_outgoing() not setting up the configured
endpoint id as the new channel's caller id. This is why party A got
default connected line information.
* Made update_initial_connected_line() use the channel's CALLERID(id)
information. The core, app_dial, or predial routine may have filled in or
changed the endpoint caller id information.
* Fixed chan_pjsip_new() not setting the full party id information
available on the caller id and ANI party id. This includes the configured
callerid_tag string and other party id fields.
* Fixed accessing channel party id information without the channel lock
held.
* Fixed using the effective connected line id without doing a deep copy
outside of holding the channel lock. Shallow copy string pointers can
become stale if the channel lock is not held.
* Made queue_connected_line_update() also update the channel's
CALLERID(id) information. Moving the channel to another bridge would need
the information there for the new bridge peer.
* Fixed off nominal memory leak in update_incoming_connected_line().
* Added pjsip.conf callerid_tag string to party id information from
enabled trust_inbound endpoint in caller_id_incoming_request().
AFS-98 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/3913/
........
Merged revisions 421400 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421403 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
be set.
Since the PJSIP INVITE session module is invoked before any session supplements it was
possible for it to handle a redirect before the res_pjsip_diversion module interpreted
and set redirecting information on the channel. This would cause the redirecting
information to get lost.
This patch ensures that session supplements are *always* invoked before a redirect occurs
by explicitly calling them in the redirect handler.
Review: https://reviewboard.asterisk.org/r/3850/
........
Merged revisions 419764 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419766 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
"module show" .
ASTERISK-23919 #close
Reported by Malcolm Davenport
Review: https://reviewboard.asterisk.org/r/3802
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419592 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
In the old times media formats were represented using a bit field. This was
fast but had a few limitations.
1. Asterisk was limited in how many formats it could handle.
2. Formats, being a bit field, could not include any attribute information.
A format was strictly its type, e.g., "this is ulaw".
This was changed in Asterisk 10 (see
https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal for
notes on that work) which led to the creation of the ast_format structure.
This structure allowed Asterisk to handle attributes and bundle information
with a format.
Additionally, ast_format_cap was created to act as a container for multiple
formats that, together, formed the capability of some entity. Another
mechanism was added to allow logic to be registered which performed format
attribute negotiation. Everywhere throughout the codebase Asterisk was
changed to use this strategy.
Unfortunately, in software, there is no free lunch. These new capabilities
came at a cost.
Performance analysis and profiling showed that we spend an inordinate
amount of time comparing, copying, and generally manipulating formats and
their related structures. Basic prototyping has shown that a reasonably
large performance improvement could be made in this area. This patch is the
result of that project, which overhauled the media format architecture
and its usage in Asterisk to improve performance.
Generally, the new philosophy for handling formats is as follows:
* The ast_format structure is reference counted. This removed a large amount
of the memory allocations and copying that was done in prior versions.
* In order to prevent race conditions while keeping things performant, the
ast_format structure is immutable by convention and lock-free. Violate this
tenet at your peril!
* Because formats are reference counted, codecs are also reference counted.
The Asterisk core generally provides built-in codecs and caches the
ast_format structures created to represent them. Generally, to prevent
inordinate amounts of module reference bumping, codecs and formats can be
added at run-time but cannot be removed.
* All compatibility with the bit field representation of codecs/formats has
been moved to a compatibility API. The primary user of this representation
is chan_iax2, which must continue to maintain its bit-field usage of formats
for interoperability concerns.
* When a format is negotiated with attributes, or when a format cannot be
represented by one of the cached formats, a new format object is created or
cloned from an existing format. That format may have the same codec
underlying it, but is a different format than a version of the format with
different attributes or without attributes.
* While formats are reference counted objects, the reference count maintained
on the format should be manipulated with care. Formats are generally cached
and will persist for the lifetime of Asterisk and do not explicitly need
to have their lifetime modified. An exception to this is when the user of a
format does not know where the format came from *and* the user may outlive
the provider of the format. This occurs, for example, when a format is read
from a channel: the channel may have a format with attributes (hence,
non-cached) and the user of the format may last longer than the channel (if
the reference to the channel is released prior to the format's reference).
For more information on this work, see the API design notes:
https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite
Finally, this work was the culmination of a large number of developer's
efforts. Extra thanks goes to Corey Farrell, who took on a large amount of the
work in the Asterisk core, chan_sip, and was an invaluable resource in peer
reviews throughout this project.
There were a substantial number of patches contributed during this work; the
following issues/patch names simply reflect some of the work (and will cause
the release scripts to give attribution to the individuals who work on them).
Reviews:
https://reviewboard.asterisk.org/r/3814
https://reviewboard.asterisk.org/r/3808
https://reviewboard.asterisk.org/r/3805
https://reviewboard.asterisk.org/r/3803
https://reviewboard.asterisk.org/r/3801
https://reviewboard.asterisk.org/r/3798
https://reviewboard.asterisk.org/r/3800
https://reviewboard.asterisk.org/r/3794
https://reviewboard.asterisk.org/r/3793
https://reviewboard.asterisk.org/r/3792
https://reviewboard.asterisk.org/r/3791
https://reviewboard.asterisk.org/r/3790
https://reviewboard.asterisk.org/r/3789
https://reviewboard.asterisk.org/r/3788
https://reviewboard.asterisk.org/r/3787
https://reviewboard.asterisk.org/r/3786
https://reviewboard.asterisk.org/r/3784
https://reviewboard.asterisk.org/r/3783
https://reviewboard.asterisk.org/r/3778
https://reviewboard.asterisk.org/r/3774
https://reviewboard.asterisk.org/r/3775
https://reviewboard.asterisk.org/r/3772
https://reviewboard.asterisk.org/r/3761
https://reviewboard.asterisk.org/r/3754
https://reviewboard.asterisk.org/r/3753
https://reviewboard.asterisk.org/r/3751
https://reviewboard.asterisk.org/r/3750
https://reviewboard.asterisk.org/r/3748
https://reviewboard.asterisk.org/r/3747
https://reviewboard.asterisk.org/r/3746
https://reviewboard.asterisk.org/r/3742
https://reviewboard.asterisk.org/r/3740
https://reviewboard.asterisk.org/r/3739
https://reviewboard.asterisk.org/r/3738
https://reviewboard.asterisk.org/r/3737
https://reviewboard.asterisk.org/r/3736
https://reviewboard.asterisk.org/r/3734
https://reviewboard.asterisk.org/r/3722
https://reviewboard.asterisk.org/r/3713
https://reviewboard.asterisk.org/r/3703
https://reviewboard.asterisk.org/r/3689
https://reviewboard.asterisk.org/r/3687
https://reviewboard.asterisk.org/r/3674
https://reviewboard.asterisk.org/r/3671
https://reviewboard.asterisk.org/r/3667
https://reviewboard.asterisk.org/r/3665
https://reviewboard.asterisk.org/r/3625
https://reviewboard.asterisk.org/r/3602
https://reviewboard.asterisk.org/r/3519
https://reviewboard.asterisk.org/r/3518
https://reviewboard.asterisk.org/r/3516
https://reviewboard.asterisk.org/r/3515
https://reviewboard.asterisk.org/r/3512
https://reviewboard.asterisk.org/r/3506
https://reviewboard.asterisk.org/r/3413
https://reviewboard.asterisk.org/r/3410
https://reviewboard.asterisk.org/r/3387
https://reviewboard.asterisk.org/r/3388
https://reviewboard.asterisk.org/r/3389
https://reviewboard.asterisk.org/r/3390
https://reviewboard.asterisk.org/r/3321
https://reviewboard.asterisk.org/r/3320
https://reviewboard.asterisk.org/r/3319
https://reviewboard.asterisk.org/r/3318
https://reviewboard.asterisk.org/r/3266
https://reviewboard.asterisk.org/r/3265
https://reviewboard.asterisk.org/r/3234
https://reviewboard.asterisk.org/r/3178
ASTERISK-23114 #close
Reported by: mjordan
media_formats_translation_core.diff uploaded by kharwell (License 6464)
rb3506.diff uploaded by mjordan (License 6283)
media_format_app_file.diff uploaded by kharwell (License 6464)
misc-2.diff uploaded by file (License 5000)
chan_mild-3.diff uploaded by file (License 5000)
chan_obscure.diff uploaded by file (License 5000)
jingle.diff uploaded by file (License 5000)
funcs.diff uploaded by file (License 5000)
formats.diff uploaded by file (License 5000)
core.diff uploaded by file (License 5000)
bridges.diff uploaded by file (License 5000)
mf-codecs-2.diff uploaded by file (License 5000)
mf-app_fax.diff uploaded by file (License 5000)
mf-apps-3.diff uploaded by file (License 5000)
media-formats-3.diff uploaded by file (License 5000)
ASTERISK-23715
rb3713.patch uploaded by coreyfarrell (License 5909)
rb3689.patch uploaded by mjordan (License 6283)
ASTERISK-23957
rb3722.patch uploaded by mjordan (License 6283)
mf-attributes-3.diff uploaded by file (License 5000)
ASTERISK-23958
Tested by: jrose
rb3822.patch uploaded by coreyfarrell (License 5909)
rb3800.patch uploaded by jrose (License 6182)
chan_sip.diff uploaded by mjordan (License 6283)
rb3747.patch uploaded by jrose (License 6182)
ASTERISK-23959 #close
Tested by: sgriepentrog, mjordan, coreyfarrell
sip_cleanup.diff uploaded by opticron (License 6273)
chan_sip_caps.diff uploaded by mjordan (License 6283)
rb3751.patch uploaded by coreyfarrell (License 5909)
chan_sip-3.diff uploaded by file (License 5000)
ASTERISK-23960 #close
Tested by: opticron
direct_media.diff uploaded by opticron (License 6273)
pjsip-direct-media.diff uploaded by file (License 5000)
format_cap_remove.diff uploaded by opticron (License 6273)
media_format_fixes.diff uploaded by opticron (License 6273)
chan_pjsip-2.diff uploaded by file (License 5000)
ASTERISK-23966 #close
Tested by: rmudgett
rb3803.patch uploaded by rmudgetti (License 5621)
chan_dahdi.diff uploaded by file (License 5000)
ASTERISK-24064 #close
Tested by: coreyfarrell, mjordan, opticron, file, rmudgett, sgriepentrog, jrose
rb3814.patch uploaded by rmudgett (License 5621)
moh_cleanup.diff uploaded by opticron (License 6273)
bridge_leak.diff uploaded by opticron (License 6273)
translate.diff uploaded by file (License 5000)
rb3795.patch uploaded by rmudgett (License 5621)
tls_fix.diff uploaded by mjordan (License 6283)
fax-mf-fix-2.diff uploaded by file (License 5000)
rtp_transfer_stuff uploaded by mjordan (License 6283)
rb3787.patch uploaded by rmudgett (License 5621)
media-formats-explicit-translate-format-3.diff uploaded by file (License 5000)
format_cache_case_fix.diff uploaded by opticron (License 6273)
rb3774.patch uploaded by rmudgett (License 5621)
rb3775.patch uploaded by rmudgett (License 5621)
rtp_engine_fix.diff uploaded by opticron (License 6273)
rtp_crash_fix.diff uploaded by opticron (License 6273)
rb3753.patch uploaded by mjordan (License 6283)
rb3750.patch uploaded by mjordan (License 6283)
rb3748.patch uploaded by rmudgett (License 5621)
media_format_fixes.diff uploaded by opticron (License 6273)
rb3740.patch uploaded by mjordan (License 6283)
rb3739.patch uploaded by mjordan (License 6283)
rb3734.patch uploaded by mjordan (License 6283)
rb3689.patch uploaded by mjordan (License 6283)
rb3674.patch uploaded by coreyfarrell (License 5909)
rb3671.patch uploaded by coreyfarrell (License 5909)
rb3667.patch uploaded by coreyfarrell (License 5909)
rb3665.patch uploaded by mjordan (License 6283)
rb3625.patch uploaded by coreyfarrell (License 5909)
rb3602.patch uploaded by coreyfarrell (License 5909)
format_compatibility-2.diff uploaded by file (License 5000)
core.diff uploaded by file (License 5000)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This corrects two issues with the extra field information in Asterisk
12+ in channel event logs.
It is possible to inject custom values into the dialstatus provided by
ast_channel_dial_type() Stasis messages that fall outside the
enumeration allowed for the DIALSTATUS channel variable. CEL now
filters for the allowed values and ignores other values.
The "hangupsource" extra field key is always blank if the far end
channel is a chan_pjsip channel. This is because the hangupsource is
never set for the pjsip channel driver. This change sets the
hangupsource whenever a hangup is queued for chan_pjsip channels.
This corrects an issue with the pjsip channel driver where the
hangupcause information was not being set properly.
Review: https://reviewboard.asterisk.org/r/3690/
........
Merged revisions 418071 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This small patch adds a debug level 3 statement indicating how a session
refresh is being sent - either as a re-INVITE or as an UPDATE - and where
the session refresh is going.
........
Merged revisions 415115 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415116 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Simply enabling PJSIP to negotiage a video codec (e.g., h264) would leak
video RTP ports if the codec were not negotiated by an incoming call.
* Made add_sdp_streams() associate the handler with the media stream if
the handler handled the media stream. Otherwise, when the
ast_sip_session_media object was destroyed it didn't know how to clean up
the RTP resources.
* Fixed sdp_requires_deferral() associating the handler with the media
stream when deciding if the SDP processing needs to be deferred for T.38.
Like the leaked video RTP ports, the T.38 handler needs to clean up
allocated resources from deciding if SDP processing needs to be deffered.
* Cleaned up some dead code in handle_incoming_sdp() and
sdp_requires_deferral().
ASTERISK-23721 #close
Reported by: cervajs
Review: https://reviewboard.asterisk.org/r/3571/
........
Merged revisions 414749 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414750 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
The Test Suite caught a few problems, undoing until those are resolved
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This patch fixes issues with direct media bridges that occur after a blind
transfer. These issues were caught by the (currently failing)
pjsip/transfers/blind_transfer/caller_direct_media test.
The test currently fails primarily for two reasons:
(1) When Bob and Charlie (the transfer target and the transfer destination)
enter a bridge together, the framehook remains on the transfer target
channel until both channels are in the bridge. As it consumes voice frames,
the initial bridge type is a simple bridge. The framehook is removed when
both channels are in the bridge; however, this does not currently cause the
bridging framework to re-evaluate the bridge. This patch adds a
AST_SOFTHANGUP_UNBRIDGE poke to the transfer target channel when a
framehook is removed so the bridge can re-evaluate itself.
(2) When a channel leaves a native RTP bridge, it may be leaving due to being
hung up. Sending a re-INVITE to a channel that is about to be hung up is
not nice - in fact, there's a good chance we'll send the BYE request before
the channel has had a chance to send back a 200 OK. To be somewhat nicer,
this patch adds a function to channel.h that allows the bridging framework
to query for exactly why a channel is leaving a bridge via the channel's
soft hangup flags. This allows it to only send the re-INVITE if there's a
chance the channel will survive the native bridging experience.
Review: https://reviewboard.asterisk.org/r/3535/
........
Merged revisions 414122 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414123 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
AST-1363
Review: https://reviewboard.asterisk.org/r/3478/
........
Merged revisions 413117 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413118 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
* Remove unused RAII_VAR() declarations. The compiler cannot catch these
because the cleanup function "references" the unused variable. Some
actually allocated and released resources that were never used.
* Fixed some whitespace issues in stasis_bridges.c.
........
Merged revisions 412399 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412400 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This change passes options to the UAS creation function. This in turn
sets up 100rel and session timer properties on the incoming session.
Reported by Julian Russell on asterisk-users mailing list.
........
Merged revisions 409287 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409288 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
The core may (depending on circumstances) request a single codec on outgoing
calls. Many channel drivers ignore or treat this as a suggestion while still
including configured codecs. The res_pjsip_session logic treated this as
an explicit request, leaving out other configured codecs.
This change makes res_pjsip_session behave like other channel driver and simply
adds the requested codec to the list.
(closes issue ASTERISK-23082)
Reported by: xrobau
Review: https://reviewboard.asterisk.org/r/3140/
........
Merged revisions 406489 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406490 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This adds Path support to chan_pjsip in res_pjsip_path.c with minimal
additions in res_pjsip_registrar.c to store the path and additions in
res_pjsip_outbound_registration.c to enable advertisement of path
support to registrars and intervening proxies.
Path information is stored on contacts and is enabled via Address of
Record (AoRs) and Registration configuration sections.
While adding path support, it became necessary to be able to add SIP
supplements that handled messages outside of sessions, so a framework
for handling these types of hooks was added in parallel to the
already-existing session supplements and several senders of
out-of-dialog requests were refactored as a result.
(closes issue ASTERISK-21084)
Review: https://reviewboard.asterisk.org/r/3050/
........
Merged revisions 405565 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405566 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Review: https://reviewboard.asterisk.org/r/3106/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This corrects the ao2_iterator opacity violations in
res_pjsip_session.c by adding a global function to get the number of
elements inside the container hidden behind the iterator.
(closes issue ASTERISK-23053)
Review: https://reviewboard.asterisk.org/r/3111/
Reported by: Richard Mudgett
........
Merged revisions 405253 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405254 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Channel creation in Asterisk is broken up into two steps: requesting and calling.
In some cases a channel may be requested but never called. This happens in the
ChanIsAvail dialplan application for determining if something is reachable or
not. The PJSIP channel driver did not take this situation into account and
attempted to end a session that was never called out on.
The code now checks the session state to determine if the session has been
called out on and if not terminates it instead of ending it.
(closes issue ASTERISK-23074)
Reported by: Kilburn
........
Merged revisions 404652 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404653 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
authentication.
The process for resending an INVITE with authentication involves restarting the UAC
session. We were incorrectly passing in that a new offer is being sent, causing the
SDP negotiation to get into a (technically speaking) funky state.
........
Merged revisions 404369 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404370 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Newer versions of PJSIP have changed to using a flag for the
PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE instead of a define. This adds a
configure check to detect the presence of the flag and use it if found.
........
Merged revisions 403329 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403330 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
Merged revisions 403271 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
The action taken when a redirect occurs is now configurable on a
per-endpoint basis. The redirect can either be treated as a redirect
to a local extension, to a URI that is dialed through the Asterisk
core, or to a URI that is dialed within PJSIP itself.
(closes issue ASTERISK-21710)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2963/
........
Merged revisions 403207 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403208 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
The direct media format capabilities are always allocated in
ast_sip_session_alloc and were not freed in the session destructor. Whoops.
(This being the third whoops caught by Scott and Nitesh's valgrind work for
the Asterisk Test Suite. Nifty!)
........
Merged revisions 402968 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402969 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
The INVITE session state callback wrongly assumes that a session will always exist, but
when rapidly terminating the session this assumption goes out the window. As all handler
code for the INVITE session state callback requires the session it will now just exit
immediately if no session exists.
(closes issue ASTERISK-22668)
Reported by: John Bigelow
........
Merged revisions 400872 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
external media address.
........
Merged revisions 400510 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400511 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Channel snapshots have string representations of the channel's native formats.
Prior to this change, the format strings were re-created on ever channel snapshot
creation. Since channel native formats rarely change, this was very wasteful.
Now, string representations of formats may optionally be stored on the ast_format_cap
for cases where string representations may be requested frequently. When formats
are altered, the string cache is marked as invalid. When strings are requested, the
cache validity is checked. If the cache is valid, then the cached strings are copied.
If the cache is invalid, then the string cache is rebuilt and copied, and the cache
is marked as being valid again.
Review: https://reviewboard.asterisk.org/r/2879
........
Merged revisions 400356 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400363 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Due to the asynchronous design of the PJMEDIA SDP negotiator it was possible for
the SDP to be negotiated *after* a channel was created and after it was being wait
on by an application. It is only after negotiation occurs that the file descriptors
for RTP are placed on the channel. Since the channel was already being waited on
these file descriptors were not monitored, causing incoming media to never be read.
This change wakes up any application waiting on the channel so that added file
descriptors end up being monitored.
(closes issue AST-1227)
Reported by: John Bigelow
........
Merged revisions 400256 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400257 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
potentially initialize it at the same time.
........
Merged revisions 400245 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400246 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
There was a collision of mod_data use on the transaction between using a nat
hook and an session response callback. During state change it was assumed
what was in the mod_data was nothing or the response callback. However, it
was possible for it to also contain a nat hook thus resulting in a bad cast
and a crash.
Added the ability to store multiple data elements in mod_data via a hash table.
In this instance, mod_data now stores a hash table of the two values that can
be retrieved using an associated string key.
(closes issue ASTERISK-22394)
Reported by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/2843/
........
Merged revisions 399990 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399991 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
res_pjsip and res_pjsip_session.
During load time in res_pjsip if an error occurred the operation would attempt to rollback all
operations done during load. This is not permitted by PJSIP as it will assert if the operation has
not been done. This fix changes the code so it will only rollback what has been initialized already.
Further changes also prevent res_pjsip and res_pjsip_session from being unloaded. This is due to
limitations within PJSIP itself. The library environment can only be changed to a certain extent
and does not provide the ability, currently, to deinitialize certain required functionality.
(closes issue ASTERISK-22474)
Reported by: Corey Farrell
........
Merged revisions 399624 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399625 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
(closes issue AST-1207)
reported by John Bigelow
Review: https://reviewboard.asterisk.org/r/2842
........
Merged revisions 399083 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
RFC 5407 section 3.1.2 details a scenario where a UAC sends
a CANCEL at the same time that a UAS sends a 200 OK for the
INVITE that the UAC is canceling. When this occurs, it is the
role of the UAC to immediately send a BYE to terminate
the call.
This scenario was reproducible by have a Digium phone with two lines
place a call to a second phone that forwarded the call to the second
line on the original phone. The Digium phone, upon realizing that it
was connecting to itself, would attempt to cancel the call. The timing
of this happened to trigger the aforementioned race condition about
80% of the time. Asterisk was not doing its job of sending a BYE
when receiving a 200 OK on a cancelled INVITE. The result was that
the ast_channel structure was destroyed but the underlying SIP
session, as well as the PJSIP inv_session and dialog, were still
alive. Attempting to perform an action such as a transfer, once in
this state, would result in Asterisk crashing.
The circumstances are now detected properly and the session is ended
as recommended in RFC 5407.
(closes issue AST-1209)
reported by John Bigelow
........
Merged revisions 397945 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397956 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395938 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
The general gist is to have a clear boundary between old SIP stuff
and new SIP stuff by having the word "SIP" for old stuff and "PJSIP"
for new stuff. Here's a brief rundown of the changes:
* The word "Gulp" in dialstrings, functions, and CLI commands is now
"PJSIP"
* chan_gulp.c is now chan_pjsip.c
* Function names in chan_gulp.c that were "gulp_*" are now "chan_pjsip_*"
* All files that were "res_sip*" are now "res_pjsip*"
* The "res_sip" directory is now "res_pjsip"
* Files in the "res_pjsip" directory that began with "sip_*" are now "pjsip_*"
* The configuration file is now "pjsip.conf" instead of "res_sip.conf"
* The module info for all PJSIP-related files now uses "PJSIP" instead of "SIP"
* CLI and AMI commands created by Asterisk's PJSIP modules now have "pjsip" as
the starting word instead of "sip"
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395764 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|