Age | Commit message (Collapse) | Author |
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Prior too this patch, the P-Asserted-Identity header would include anonymous
caller id information which seems to go against the point of the
P-Asserted-Identity header. Now the real caller ID information will be
included in this header. Also, no privacy header would be included.
This patch adds 'Privacy: id' to outgoing SIP messages that include the
P-Asserted-Identity header.
(closes issue AST-1301)
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If update_provisional_keepalive() is called while
send_provisional_keepalive_full() is waiting on the PVT lock, then
pvt->provisional_keepalive_sched_id will be changed to a new sched_id
value by update_provisional_keepalive(), but that new sched_id then may
be overwritten with -1 by send_provisional_keepalive_full(), killing
the pvt's reference to a schedule and "leaking" the reference.
(closes issue ASTERISK-22079)
Review: https://reviewboard.asterisk.org/r/3368/
Reported by: Jamuel Starkey, Matteo, Leif Madsen, Steve Davies
Patches:
provisional_keepalive_fix.diff uploaded by Steve Davies (license 5012)
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to restricted.
(closes issue ASTERISK-20841)
Reported by: Kelly Goedert
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This change allows chan_sip to avoid creation of the channel and
consumption of associated file descriptors altogether if the inbound
request is going to be rejected anyway.
(closes issue ASTERISK-23373)
Reported by: Corey Farrell
Patches:
chan_sip-earlier-st-1.8.patch uploaded by Corey Farrell (license 5909)
chan_sip-earlier-st-11.patch uploaded by Corey Farrell (license 5909)
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Release monlock before calling pthread_join. This ensures do_monitor
cannot freeze by locking monlock during module unload.
(closes issue ASTERISK-21406)
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/3284/
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errors.
(issue ASTERISK-23120)
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Much needed was a way to assign id to objects on creation, and
much change was necessary to accomplish it. Channel uniqueids
and linkedids are split into separate string and creation time
components without breaking linkedid propgation. This allowed
the uniqueid to be specified by the user interface - and those
values are now carried through to channel creation, adding the
assignedids value to every function in the chain including the
channel drivers. For local channels, the second channel can be
specified or left to default to a ;2 suffix of first. In ARI,
bridge, playback, and snoop objects can also be created with a
specified uniqueid.
Along the way, the args order to allocating channels was fixed
in chan_mgcp and chan_gtalk, and linkedid is no longer lost as
masquerade occurs.
(closes issue ASTERISK-23120)
Review: https://reviewboard.asterisk.org/r/3191/
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When a static realtime peer with qualify=yes is loaded, Asterisk will fail to
send an OPTIONS request due to the lastms being equal to 0. This results in
the peer being unable to receive calls from Asterisk because the status is
permanently UNKNOWN.
This patch allows an OPTIONS request to be sent during module load by
ignoring the lastms value on startup only.
Review: https://reviewboard.asterisk.org/r/3294/
(closes issue ASTERISK-17523)
Reported by: Maciej Krajewski
Tested by: wushumasters
patches:
realtime_fix_11.7.0.txt uploaded by Trevor Peirce (license 6112)
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functions conforming to other channel drivers. Do not forget auto-detected and user-selected phone settings on 'unistim reload'
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(closes issue ASTERISK-23406)
Reported by: ibercom
Tested by: ibercom
Patches:
asterisk-11.patch uploaded by ibercom
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* Add precautionary p->owner checks in sip_hangup(), get_refer_info(),
get_also_info(), and interpret_t38_parameters().
* Simplify some tangled logic in get_refer_info(), get_also_info(), and
add_rpid().
* Removed some dead code in handle_request_invite().
(closes issue ASTERISK-23323)
Reported by: Walter Doekes
Patches:
issueA23323-more_p_owner_checks-1.8.x.patch (license #5674) uploaded by wdoekes (modified)
issueA23323-more_p_owner_checks-11.x.patch (license #5674) uploaded by wdoekes (modified)
issueA23323-more_p_owner_checks-12.x.patch (license #5674) uploaded by wdoekes (modified)
issueA23323-more_p_owner_checks-trunk.patch (license #5674) uploaded by wdoekes (modified)
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* Fix crash in ast_channel_hangupcause_set() because p->owner not checked
before calling. Regression introduced by the fix for ASTERISK-22621.
(closes issue ASTERISK-23135)
Reported by: OK
(issue ASTERISK-23323)
Reported by: Walter Doekes
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When accidentally compiling against a wrong version of
pjsip headers with a different pjsip_inv_session size,
the invite_tsx structure could be null in the answer()
function. This led to a crash because it attempted to
send the session response with an uninitialized packet
pointer. This patch presets packet to null and adds a
diagnostic log message to explain why the call fails.
Review: https://reviewboard.asterisk.org/r/3267/
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support.
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Fix regression caused by ASTERISK-22582. Empty Route
headers were added when the route had a single strict
hop.
(closes issue ASTERISK-23306)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3236/
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This patch moves setting SIP_DEFER_BY_ON_TRANSFER prior to calling
ast_bridge_transfer_blind. This prevents a BYE from being sent prior to the
NOTIFY request that informs the transferor if the transfer succeeded or failed.
This patch also clears said flag from the off nominal NOTIFY paths in the
local_attended_transfer code, as once we've sent the NOTIFY request it is safe
to send by the BYE request.
This was caught by the blind-transfer-accountcode test in the Asterisk Test
Suite.
(closes issue ASTERISK-23290)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3214/
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* Move route code to sip/route.c + sip/include/route.h
* Rename functions to sip_route_*
* Replace ad-hoc list code with macro's from linkedlists.h
* Create sip_route_process_header() to processes Path and Record-Route headers
(previously done with different code in build_route and build_path)
* Add use of const where possible
* Move struct uriparams, struct contact and contactliststruct from sip.h to
reqresp_parser.h. sip/route.c uses reqresp_parser.h but not sip.h, this was
a problem. These moved declares are not used outside of reqresp_parser.
* While modifying reqprep() the lack of {} caused me trouble. I added them.
* Code outside route.c treats sip_route as an opaque structure, using macro's
or procedures for all access.
(closes issue ASTERISK-22582)
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/3173/
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Establishing an IAX2 call between Asterisk v1.4 and v1.8 (or later)
results in an unexpected call disconnect. The problem happens because
newer values in the enum ast_control_frame_type are not consistent between
the branch versions of Asterisk.
For example:
1) v1.4 calls v1.8 (or later) using IAX2
2) v1.8 answers and sends a connected line update control frame. (on v1.8
AST_CONTROL_CONNECTED_LINE = 22)
3) v1.4 receives the control frame as an end-of-q (on v1.4
AST_CONTROL_END_OF_Q = 22)
4) v1.4 disconnects the call once the receive queue becomes empty.
Several things are done by this patch to fix the problem and attempt to
prevent it from happening again in the future:
* Added a warning at the definition of enum ast_control_frame_type about
how to add new control frame values.
* Made block sending and receiving control frames that have no reason to
go over the wire.
* Extended the connectedline iax.conf parameter to also include the
redirecting information updates.
* Updated the connectedline iax.conf parameter documentation to include a
notice that the parameter must be "no" when the peer is an Asterisk v1.4
instance.
(closes issue AST-1302)
Review: https://reviewboard.asterisk.org/r/3174/
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Locking issues in skinny when picking up a call that doesn't exist. Cleaned
up sub locking by fully removing and using the chan lock instead. Also
changed ast_call_pickup to check whether chan was masq'd.
(closes issue ASTERISK-23249)
Reported by: wedhorn
Tested by: snuffy, myself
Patches:
skinny-locking01.diff uploaded by wedhorn (license 5019)
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Return before chan is possibly unlocked a second time when hanging up
a channel in SUBSTATE_OFFHOOK.
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This change allows chan_sip to decline individual image streams over
unsupported transports in the SDP of the 200 response. Previously,
an image stream offer with RTP/AVP as the transport would cause
chan_sip to respond with a 488.
(closes issue ASTERISK-22988)
Reported by: adomjan
Original patch by: adomjan
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This restricts direct usage of global oseq so that all accesses are
locked and threads are not racing to get oseq values that they did not
claim.
This also fixes a build error in res_pktccops under dev mode.
(closes issue ASTERISK-23100)
Reported by: adomjan
Patch by: adomjan
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Bad offset in reading second or more fragment of skinny packets. Fixed
to offset by char (single byte) rather than size of req.
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is valid.
* Also simplified some subddress handling code.
(closes issue ASTERISK-23008)
Reported by: Michael Cargile
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Logging from the skinny session loop was providing some incorrect reasons
for exiting the loop. Cleaned up messages and handling so correct reason
displayed.
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Previously PJSIP did not track hold/unhold and it would always simply be
'inuse'. This patch fixes that.
review: https://reviewboard.asterisk.org/r/3129/
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Existing code would do a full device restart when "skinny reset device"
was entered at the CLI and do a reset when "skinny reset device restart"
entered.
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issues 23061,23028,23046,23027
Fixes typos of "transfered" instead of "transferred" in various code. Fixes incorrect gosub param help text for app_queue.
Fixes Asterisk man pages containing unquoted minus signs. Adds note about the "textsupport" option in sip.conf.sample.
(issue ASTERISK-23061)
(issue ASTERISK-23028)
(issue ASTERISK-23046)
(issue ASTERISK-23027)
(closes issue ASTERISK-23061)
(closes issue ASTERISK-23028)
(closes issue ASTERISK-23046)
(closes issue ASTERISK-23027)
Reported by: Eugene, Jeremy Laine, Denis Pantsyrev
Patches:
transferred.patch uploaded by Jeremy Laine (license 6561)
hyphen.patch uploaded by Jeremy Laine (license 6561)
sip.conf.sample.patch uploaded by Eugene (license 6360)
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When endpoints get loaded their device state gets set to 'INVALID' because the
channel driver has not been loaded yet. Fixed by updating the device state for
every endpoint upon load of the channel driver.
(closes issue ASTERISK-23065)
Reported by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/3123/
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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/
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In ASTERISK-12117, an improvement to insure consistant local from tags
on outbound registrations resulted in an undesirable behavior - caused
by leftover unexpired sip_pvt dialogs (with the previous cseq number),
resulting in many uncessary REGISTER requests. Instead of significant
rework of transmit_register(), this change deletes the dialogs after a
200 OK response indiciating a successful registration, keeping the old
dialogs from interfering with normal operation.
(closes issue ASTERISK-22946)
Reported by: Stephan Eisvogel
Review: https://reviewboard.asterisk.org/r/3109/
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Patch updates skinny devices with a SKINNY_CONNECTED callstate if an
inbound ringing or callwaiting call is answered elsewhere.
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Error "unused variable i in dahdi_create_channel_range" when compiling
in dev-mode. Small restructure to dahdi_create_channel_range to move
the for(x) loop and int i,x to a block within the IFDEF.
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When the patch to add retry-on-forbidden-response was committed, part
of the patch for chan_sip was not committed which caused the feature to
be entirely nonfunctional. This corrects the code in question.
(closes issue ASTERISK-17138)
Review: https://reviewboard.asterisk.org/r/2874
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dahdi show channels output slices the callerid (which is dnid copied over on
PRI channels). If the channel naming structures look like:
'DAHDI/i1/1408409XXXX-6'
then the output slices 1408409XXXX down to 1408409XXX. This patch just opens
it up to 15 chars so you can see the whole thing.
(closes issue ASTERISK-22918)
Reported by: outtolunc
Patches:
svn_chan_dahdi.c.format12_15.diff.txt uploaded by outtolunc (license 5198)
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Added a new 'set_var' option for ast_sip_endpoint(s). For each variable
specified that variable gets set upon creation of a pjsip channel involving
the endpoint.
(closes issue ASTERISK-22868)
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/3095/
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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
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If ignore_failed_channels is set to "true" for a channel, the channel
will continue to be configured even if configuring it has failed.
This allows Asterisk to start before all the DAHDI initialization is
done and thus not force the starting order dahdi -> asterisk.
Review: https://reviewboard.asterisk.org/r/3063/
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Removing dead code starting with ast_udptl_bridge() eliminated the code in
this change.
Note: This code has actually been dead since Asterisk v1.4 when it was
first put in.
Review: https://reviewboard.asterisk.org/r/3079/
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system.
This change is in preparation for external MWI support.
Removed code from the system for normal mailbox handling that appends
@default to the mailbox identifier if it does not have a context. The
only exception is the legacy hasvoicemail users.conf option. The legacy
option will only work for app_voicemail mailboxes. The system cannot make
any assumptions about the format of the mailbox identifer used by
app_voicemail.
chan_sip and chan_dahdi/sig_pri had the most changes because they both
tried to interpret the mailbox identifier. chan_sip just stored and
compared the two components. chan_dahdi actually used the box
information.
The ISDN MWI support configuration options had to be reworked because
chan_dahdi was parsing the box@context format to get the box number. As a
result the mwi_vm_boxes chan_dahdi.conf option was added and is documented
in the chan_dahdi.conf.sample file.
Review: https://reviewboard.asterisk.org/r/3072/
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Moved channel locking into setsubstate so that a process can complete
working on a sub before another starts changing it. The existing code
was causing some Fracks with schedule deletion.
Removed multiple rtp cleanup. Now only cleansup up once, fixing ao2
object cleanup issues.
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On session registration, if device is already reporting that it is
connected to a device, an innocuous packet (update time) is sent to
the already connected device. If the tcp connection is down, the
device will be unregistered and the new connection allowed.
Without this patch, network issues can see a situation where a device
can not reregister until after 3*timeout.
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Removed channel lock as it is now being down in ast_channel_alloc
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Original commit message by mmichelson (asterisk 12 r403311):
"This adds channel locks around calls to create channel snapshots as well
as other functions which operate on a channel and then end up
creating a channel snapshot. Functions that expect the channel to be
locked prior to being called have had their documentation updated to
indicate such."
The above was initially committed and then reverted at r403398. The problem
was found to be in core_local.c in the publish_local_bridge_message function.
The ast_unreal_lock_all function locks and adds a reference to the returned
channels and while they were being unlocked they were not being unreffed when
no longer needed. Fixed by unreffing the channels.
Also in bridge.c a lock was obtained on "other->chan", but then an attempt was
made to unlock "other" and not the previously locked channel. Fixed by
unlocking "other->chan"
(closes issue ASTERISK-22709)
Reported by: John Bigelow
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This change makes ast_channel_alloc return allocated channels
locked. By doing so no other thread can acquire, lock, and manipulate
the channel before it is completely set up.
(closes issue AST-1256)
Review: https://reviewboard.asterisk.org/r/3067/
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of "available"
(issue ASTERISK-23021)
(closes issue ASTERISK-23021)
Reported by: Jeremy Lainé
Tested by: Rusty Newton
Patches:
available.patch uploaded by Jeremy Lainé (license 6561)
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While entertaining, sizeof(buflen) is not the same as buflen. Doh.
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The change contains a slightly adjusted patch that was on the issue
(submitted by kmoore). A fix was made by adding in a bridge lock
while calling bridge_start/stop from the framehook callback. Since
the framehook callback is not called from the bridging core the bridge
is not locked, but needs to be before calling bridge_start.
(closes issue ASTERISK-22749)
Reported by: Kinsey Moore
Review: https://reviewboard.asterisk.org/r/3066/
Patches:
lock_inversion.diff uploaded by kmoore (license 6273)
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