In order to alter the Contact header on in-dialog requests and responses the
Websocket module must be attached on outgoing INVITEs. The Contact header is
modified so that the PJSIP transport layer can find and use the existing
Websocket connection based on the source IP address, port, and transport.
ASTERISK-24143 #close
Reported by: Aleksei Kulakov
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The packet structure used to receive messages was using the transport
pool. This meant that for each parsing the pool would grow accordingly.
Since memory can not be reclaimed without resetting it this would
cause the memory pool to grow and grow.
This change uses a specific memory pool for the packet structure and
resets it to a fresh state after the message has been received and
handled.
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This change enforces the transport in the Contact header for Websocket clients.
Previously a client may provide a transport of 'ws' when it is actually using
a transport of 'wss'. This would cause outgoing calls to fail as the existing
connection could not be found.
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This code originally worked around an issue within res_rtp_asterisk itself.
The wrong socket was being used for the STUN check for RTCP, causing the
port to be the same as RTP. This was subsequently fixed and the RTCP port
provided for the ICE candidate is correct and does not need to be incremented.
ASTERISK-23997 #close
Reported by: Badalian Vyacheslav
Patches:
plus1.diff submitted by Badalian Vyacheslav (license 5249)
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We need to unlock the audiohook before trying to lock
the channel, since the correct locking order is channel
then audiohook.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421882 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
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Because of the departable state of channels that enter Stasis bridges, Stasis has to
take responsibility for directing the channel to its intended after-bridge destination
if the channel moves from a Stasis bridge to a non-Stasis bridge. This change ensures
that when such a move occurs, when the channel leaves the bridging system, any after
bridge gotos are honored.
Review: https://reviewboard.asterisk.org/r/3920
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Prior to this change, the Remote-Party-ID header took the position of
"If caller name and number are not explicitly allowed, then they are private"
and P-Asserted-Identity took the position of
"Caller name and number are only private if marked explicitly so"
Now both mechanisms of conveying party identification use the former approach.
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If a user does not provide a port in the fromdomain setting, chan_sip will set
the fromdomainport to STANDARD_SIP_PORT (5060). The fromdomainport value will
then get used unilaterally in certain places. This causes issues with TLS,
where the default port is expected to be 5061.
This patch modifies chan_sip such that fromdomainport is only used if it is
not the standard SIP port; otherwise, the port from the SIP pvt's recorded
self IP address is used.
Review: https://reviewboard.asterisk.org/r/3893/
ASTERISK-24178 #close
Reported by: Elazar Broad
patches:
fromdomainport_fix.diff uploaded by Elazar Broad (License 5835)
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When issuing a POST /channels/{channel_id}/play on a channel that is not
yet answered, ARI is supposed to:
* Queue up an AST_CONTROL_PROGRESS on the channel
* Start up the playback of the media
Instead, we sneak an answer on the channel right before starting playing media.
This is due to ARI's usage of control_streamfile. This function implicitly
answers the channel (and doesn't give ARI the option to stop it). The answering
of the channel here is probably unnecessary:
* app_voicemail, by far the biggest consumer of this function, always answers
the channels anyway
* control stream file (in res_agi) and ControlPlayback probably shouldn't be
implicitly answering the channel. Answering should not be tied directly to
playing back media.
As it turns out, the answering of the channel here is pretty old:
356042 twilson if (ast_channel_state(chan) != AST_STATE_UP) {
3087 anthm res = ast_answer(chan);
180259 tilghman }
(As in, ancient?)
Note that others ran into this problem and commented about it on various
mailing lists.
Review: https://reviewboard.asterisk.org/r/3907/
ASTERISK-24229 #close
Reported by: Matt Jordan
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Trivial patch to add new lines to several files missing them. This fixes
warnings when compiling with gcc 4.1.2 on CentOS 5.
ASTERISK-24245 #close
Reported by: Shaun Ruffell
patches:
0002-Trivial-addition-of-newlines-at-end-of-three-files.patch uploaded by Shaun Ruffell (License 5417)
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This patch fixes gcc warnings that occur due to the type qualifier 'const'
being ignored on a return type of int.
ASTERISK-24246 #close
Reported by: Shaun Ruffell
patches:
0001-main-uri-Quiet-warning-about-ignored-attribute-on-re.patch uploaded by Shaun Ruffell (License 5417)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421675 65c4cc65-6c06-0410-ace0-fbb531ad65f3
On a SIP reinvite that changes media strams, the PJSIP channel driver was
flooding the log with "Asked to transmit frame type %s, while native
formats is %s" warnings.
* Fixes PJSIP not setting up translation paths when the formats change on
a reinvite. AFS-63 was effectively reintroduced because of the media
formats work. res_pjsip_sdp_rtp.c:set_caps()
* Improved the unexpected frame format WARNING message to include more
information.
* Added protective locking while altering formats on a channel. Reworked
set_format() to simplify and protect the formats under manipulation.
* Restored some code that got lost in the media_formats work.
(channel.c:set_format() and res_pjsip_sdp_rtp.c:set_caps())
AFS-137 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/3906/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421645 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This was causing the AMI show_subscriptions test in
the testsuite to fail since all subscriptions were being
seen as subscribers instead of notifiers.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421585 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This allows for set_var to override certain defaults such as caller ID and codec
values. This also fixes a test suite regression. The "set_var" test suite test attempted
to use set_var to override caller ID, but a recent change caused that to no longer work.
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When a blind transfer occurs that is forced to create a local channel
pair to satisfy the transfer request, information about the local
channel pair is not published. This adds a field to describe that
channel to the blind transfer message struct so that this information
is conveyed properly to consumers of the blind transfer message.
This also fixes a bug in which Stasis() was unable to properly identify
the channel that was replacing an existing Stasis-controlled channel
due to a blind transfer.
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3921/
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If /channels/{channelID}/continue is called on a channel that was originated
without a PBX (such as the ARI command POST channel with a stasis application
argument), the channel will not start dialplan execution. This patch will now
run the PBX out of the stasis execution if the channel doesn't currently have
an active PBX upon continuing.
ASTERISK-24043 #close
Reported by: Krandon Bruse
Review: https://reviewboard.asterisk.org/r/3917/
Patches:
stasis-continue.diff submitted by Krandon Bruse (license 6631)
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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/
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If a function fails to execute, it is most likely due to one of two reasons:
(1) The function doesn't exist or can't be read from
(2) The function is dangerous and is restricted based on the user's permissions
Currently we return allocation failure, which is incorrect. This updates the
reason code to more accurately reflect why the request failed.
ASTERISK-24215
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The same function, meetme_stasis_generate_msg, handles creating and publishing
Stasis message both when there are channels in the MeetMe conference and when
there are no channels in the conference. When the performance improvement was
made to use cached snapshots, this created a situation where Asterisk would
crash: obtaining a cached snapshot is not NULL tolerant.
This patch restores the previous implementation, which used a NULL safe set
of routines to produce a blob containing the channel snapshot (if available)
and information about the MeetMe conference.
ASTERISK-24234 #close
Reported by: Shaun Ruffell
Tested by: Shaun Ruffell
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Some distributions of Linux patch gcc to define FORTIFY_SOURCE when gcc is
executed with optimization. This "help" unfortunately results in re-definition
warnings when FORTIFY_SOURCE is later defined in Asterisk's build system. This
patch undefines FORTIFY_SOURCE prior to defining it to prevent this warning.
Review: https://reviewboard.asterisk.org/r/3912/
ASTERISK-24032 #close
Reported by: Kilburn
Tested by: Kilburn, wdoekes
patches:
1.8.diff uploaded by cloos (License 5956)
10.diff uploaded by cloos (License 5956)
11.diff uploaded by cloos (License 5956)
12.diff uploaded by cloos (License 5956)
13.diff uploaded by cloos (License 5956)
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r420934 introduced some failures in the test suite. Upon investigating, it was
discovered that differences in the way we were evaluating whether a channel was in
the process of leaving a bridge were causing some reinvites not to occur (mostly
reinvites back to Asterisk when ending a call). This patch fixes that behavioral
change.
ASTERISK-24027 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3910/
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Moving the test event raised when a file is played back (which occurred in
r421059) broke the ever loving snot out of the voicemail tests. This caused
duplicate test events to get raised, as app_voicemail and main/app were raising
events prior to call ast_streamfile. The voicemail tests did not enjoy getting
multiple events.
Since raising the playback event in ast_streamfile is far more useful to the
vast majority of tests, this patch keeps the call there and simply removes the
extraneous calls that duplicated the event.
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The res_hep_rtcp module was incorrectly including <pjsip.h>. This didn't need
to be included, as the module does not using PJPROJECT any fashion.
Unfortunately, because res_hep_rtcp did not include pjsip in its MODULEINFO as
a dependency, this also meant that res_hep_rtcp will fail to compile on a
system without PJPROJECT.
This patch removes the include.
Thanks to Damien Wedhorn for pointing this out in #asterisk-dev.
ASTERISK-24236 #close
Reported by: Damien Wedhorn, Matt Jordan
Tested by: Damien Wedhorn
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CEL typically tracks a lot of information using the unique ID of the channel.
This is typically needed due to tying events together using the linked ID of
the various channels involved in a "call", which is derived from the channel ID
of the oldest channel involved in a bridge (or in the case of a Dial, the
parent channel).
Previously, we had updated the extra fields to include the involved channel
names, but forgot to put in the unique ID. This patch corrects that error.
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If a manager or CLI user attached a mixmonitor to a call running a dynamic
bridge feature while in a bridge, the feature would be interrupted and the
channel would be forcibly kicked out of the bridge (usually ending the call
during a simple 1 to 1 call). This would also occur during any similar action
that could set the unbridge soft hangup flag, so the fix for this was to
remove unbridge from the soft hangup flags and make it a separate thing all
together.
ASTERISK-24027 #close
Reported by: mjordan
Review: https://reviewboard.asterisk.org/r/3900/
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