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When the SIP registrations were migrated to using ao2 in what was then trunk,
the explicit destruction of the registrations on module reload was removed and
not replaced with an ao2 equivalent. Debugging done by Stefan Engström, the
issue reporter, on ASTERISK-24673 confirmed that the reference in the
registry_list container was being leaked.
Since the purpose of cleanup_all_regs is to prep a registration for
destruction, this function now calls an ao2_callback function callback with the
OBJ_MULTIPLE | OBJ_NODATA | OBJ_UNLINK flags used to remove the registrations.
This cleans up each registration, and also removes it from the registration
container registry_list.
Review: https://reviewboard.asterisk.org/r/4355/
ASTERISK-24640 #close
Reported by: Max Man
ASTERISK-24673 #close
Reported by: Stefan Engström
Tested by: Stefan Engström
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across a bridge.
Calling ast_channel_bridge_peer() cannot be done while holding any channel
locks. The reported issue hit the deadlock in chan_iax2, but an audit of
the ast_channel_bridge_peer() calls found three more locations where the
same deadlock can occur.
* Made CHANNEL(peer), res_fax, and the SNMP agent not call
ast_channel_bridge_peer() with any channel locked. For CHANNEL(peer) I
had to rework the logic to not hold the channel lock.
* Made chan_iax2 no longer call ast_channel_bridge_peer(). It was done
for legacy reasons that no longer apply.
* Removed the iax.conf forcejitterbuffer option. It is now always enabled
when the jitterbuffer option is enabled. If you put a jitter buffer on a
channel it will be on the channel.
ASTERISK-24600 #close
Reported by: Jeff Collell
Review: https://reviewboard.asterisk.org/r/4342/
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Different clients react differently to being told that a blind transfer
has failed. Some will simply send a BYE and be done with it. Others will
attempt to reinvite themselves back onto the call.
In the latter case, we were creating a new channel and then leaving it to
sit forever doing nothing. With this code change, that new channel will
not be created and the dialog with the transferring channel will be cleaned
up properly.
ASTERISK-24624 #close
Reported by Zane Conkle
Review: https://reviewboard.asterisk.org/r/4339
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The 'pjsip_get_dest_info' function is used to determine if the signaling transport
of the dialog is secure or not. This function was added in PJSIP 2.3 and does not
exist in earlier versions.
This configure check allows Asterisk to build and run with older versions at the
loss of the 'secure' argument for the PJSIP CHANNEL dialplan function. Usage of
this argument will require upgrading to PJSIP 2.3.
ASTERISK-24665 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4329/
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instead.
ASTERISK-24049
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Follow-up issue to -r430435 from reviewboard review.
ASTERISK-24049
Review: https://reviewboard.asterisk.org/r/4315/
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* Made the following AMI actions use list API calls for consistency:
Agents
BridgeInfo
BridgeList
BridgeTechnologyList
ConfbridgeLIst
ConfbridgeLIstRooms
CoreShowChannels
DAHDIShowChannels
DBGet
DeviceStateList
ExtensionStateList
FAXSessions
Hangup
IAXpeerlist
IAXpeers
IAXregistry
MeetmeList
MeetmeListRooms
MWIGet
ParkedCalls
Parkinglots
PJSIPShowEndpoint
PJSIPShowEndpoints
PJSIPShowRegistrationsInbound
PJSIPShowRegistrationsOutbound
PJSIPShowResourceLists
PJSIPShowSubscriptionsInbound
PJSIPShowSubscriptionsOutbound
PresenceStateList
PRIShowSpans
QueueStatus
QueueSummary
ShowDialPlan
SIPpeers
SIPpeerstatus
SIPshowregistry
SKINNYdevices
SKINNYlines
Status
VoicemailUsersList
* Incremented the AMI version to 2.7.0.
* Changed astman_send_listack() to not use the listflag parameter and
always set the value to "Start" so the start capitalization is consistent.
i.e., The FAXSessions used "Start" while the rest of the system used
"start". The corresponding complete event always used "Complete".
* Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the
AMI ActionID for all of its list events.
* Fixed off-nominal AMI protocol error in manager_bridge_info(),
manager_parking_status_single_lot(), and
manager_parking_status_all_lots(). Use of astman_send_error() after
responding to the original AMI action request violates the action response
pattern by sending two responses.
* Fixed minor protocol error in action_getconfig() when no requested
categories are found. Each line needs to be formatted as "Header: text".
* Fixed off-nominal memory leak in manager_build_parked_call_string().
* Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail().
ASTERISK-24049 #close
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/4315/
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The PJSIP_AOR dialplan function allows inspection of configured AORs including
what contacts are currently bound to them.
The PJSIP_CONTACT dialplan function allows inspection of contacts in existence.
These can include both externally added (by way of registration) or permanent
ones.
ASTERISK-24341
Reported by: xrobau
Review: https://reviewboard.asterisk.org/r/4308/
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Given the following scenario:
* Three SIP phones (A, B, C), all communicating via a proxy with Asterisk
* A call is established between A and B. B performs a SIP attended transfer of
A to C. B sets the call on hold (A is hearing MOH) and dials the extension of
C. While phone C is ringing, B transfers the call (that is, what we typically
call a 'blond transfer').
* When the transfer completes, A hears the ringing of phone C, while B is idle.
In the SIP messaging for the above scenario, a REFER request is sent to
transfer the call. When "sendrpid=yes" is set in sip.conf, Asterisk may send an
UPDATE request to phone C to update party information. This update is sent
directly to phone C, not through the intervening proxy. This has the unfortunate
side effect of providing route information, which is then set on the sip_pvt
structure for C. If someone (e.g. B) is trying to get the call back (through a
directed pickup), Asterisk will send a CANCEL request to C. However, since we
have now updated the route set, the CANCEL request will be sent directly to C
and not through the proxy. The phone ignores this CANCEL according to RFC3261
(Section 9.1).
This patch updates reqprep such that the route is not updated if an UPDATE
request is being sent while the INVITE state is INV_PROCEEDING or
INV_EARLY_MEDIA. This ensures that a subsequent CANCEL request is still sent
to the correct location.
Review: https://reviewboard.asterisk.org/r/4279
ASTERISK-24628 #close
Reported by: Karsten Wemheuer
patches:
issue.patch uploaded by Karsten Wemheuer (License 5930)
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When the configuration section scheme of chan_dahdi.conf is used (keyword
dahdichan instead of channel) all setvar= options are completely ignored.
No variable defined this way appears in the created DAHDI channels.
* Move the clearing of setvar values to after the deferred processing of
dahdichan.
AST-1378 #close
Reported by: Guenther Kelleter
Patch by: Guenther Kelleter
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ASTERISK-24337 #close
Reported by: Rusty Newton
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mode.
For the featdmf signaling mode the incoming MF Caller-ID information is
formatted as follows: *${CALLERID(ani2)}${CALLERID(ani)}#*${EXTEN}#
Rather than discarding the ani2 digits, populate the CALLERID(ani2) value
with what is received instead.
AST-1368 #close
Reported by: Denis Martinez
Patches:
extract_ani2_for_featdmf_v11.patch (license #5621) patch uploaded by Richard Mudgett
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Prior to this patch, we were using the PJSIP dialog's secure flag
to determine if a secure transport was being used. Unfortunately,
the dialog's secure flag was only set if a SIPS URI were in use,
as required by RFC 3261 sections 12.1.1 and 12.1.2. What we're interested
in is not dialog security, but transport security. This code change
switches to a model where we use the dialog's target URI to determine
what transport would be used to communicate, and then check if that
transport is secure.
AST-1450 #close
Reported by John Bigelow
Review: https://reviewboard.asterisk.org/r/4277
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In r413586 (1.8) various casts were added to silence gcc 4.10 warnings.
Those fixes included things like:
-out += sprintf(out, "%%%02X", (unsigned char) *ptr);
+out += sprintf(out, "%%%02X", (unsigned) *ptr);
That works for low ascii characters, but for the high range that yields
e.g. FFFFFFC3 when C3 is expected.
This changeset:
- fixes those casts to use the 'hh' unsigned char modifier instead
- consistently uses %02x instead of %2.2x (or other non-standard usage)
- adds a few 'h' modifiers in various places
- fixes a 'replcaes' typo
- dev/urandon typo (in 13+ patch)
Review: https://reviewboard.asterisk.org/r/4263/
ASTERISK-24619 #close
Reported by: Stefan27 (on IRC)
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Previously when SRTP was enabled on a channel it was not possible
to switch to T.38 as no crypto attributes would be present.
This change makes it so it is now possible. If a T.38 re-invite
comes in SRTP is terminated since in practice you can't encrypt
a UDPTL stream. Now... if we were doing T.38 over RTP (which
does exist) then we'd have a chance but almost nobody does that so
here we are.
ASTERISK-24449 #close
Reported by: Andreas Steinmetz
patches:
udptl-ignore-srtp-v2.patch submitted by Andreas Steinmetz (license 6523)
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When direct media is enabled and a pjsip channel is answered a race would occur
between the handling of the answer and bridge setup. Sometimes the media
negotiation would take place after the native bridge was setup. This resulted
in a NULL media address, which in turn resulted in Asterisk using its address
as the remote media address when sending a reinvite. This patch makes the
chan_pjsip answer handler synchronous thus alleviating the race condition (the
bridge won't start setting things up until after it returns).
ASTERISK-24563 #close
Reported by: Steve Pitts
Review: https://reviewboard.asterisk.org/r/4257/
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Given the scenario where a PJSIP channel is in a native RTP bridge with direct
media and the channel is then hung up the code will currently re-INVITE the channel
back to Asterisk and send a BYE at the same time. Many SIP implementations dislike
this greatly.
This change makes it so that if a re-INVITE transaction is in progress the BYE
is queued to occur after the completion of the transaction (be it through normal
means or a timeout).
Review: https://reviewboard.asterisk.org/r/4248/
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payload length of zero.
Frames with a payload length of 0 were incorrectly handled in res_http_websocket.
Provided a frame with a payload had been received prior it was possible for a double
free to occur. The realloc operation would succeed (thus freeing the payload) but be
treated as an error. When the session was then torn down the payload would be
freed again causing a crash. The read function now takes this into account.
This change also fixes assumptions made by users of res_http_websocket. There is no
guarantee that a frame received from it will be NULL terminated.
ASTERISK-24472 #close
Reported by: Badalian Vyacheslav
Review: https://reviewboard.asterisk.org/r/4220/
Review: https://reviewboard.asterisk.org/r/4219/
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When endpoints with direct_media enabled, behind a firewall (Asterisk on a
separate network) and were bridged sometimes Asterisk would send the ip
address of the firewall in the sdp to one of the phones in the reinvite
resulting in one way audio. When sending the reinvite Asterisk will retrieve
the media address from the associated rtp instance, but if frames were being
read this can be overwritten with another address (in this case the
firewall's). This patch ensures that Asterisk uses the original device
address when using direct media.
ASTERISK-24563
Reported by: Steve Pitts
Review: https://reviewboard.asterisk.org/r/4216/
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Prior to this patch, all Stasis subscriptions would receive a dedicated
thread for servicing published messages. In contrast, prior to r400178
(see review https://reviewboard.asterisk.org/r/2881/), the subscriptions
shared a thread pool. It was discovered during some initial work on Stasis
that, for a low subscription count with high message throughput, the
threadpool was not as performant as simply having a dedicated thread per
subscriber.
For situations where a subscriber receives a substantial number of messages
and is always present, the model of having a dedicated thread per subscriber
makes sense. While we still have plenty of subscriptions that would follow
this model, e.g., AMI, CDRs, CEL, etc., there are plenty that also fall into
the following two categories:
* Large number of subscriptions, specifically those tied to endpoints/peers.
* Low number of messages. Some subscriptions exist specifically to coordinate
a single message - the subscription is created, a message is published, the
delivery is synchronized, and the subscription is destroyed.
In both of the latter two cases, creating a dedicated thread is wasteful (and
in the case of a large number of peers/endpoints, harmful). In those cases,
having shared delivery threads is far more performant.
This patch adds the ability of a subscriber to Stasis to choose whether or not
their messages are dispatched on a dedicated thread or on a threadpool. The
threadpool is configurable through stasis.conf.
Review: https://reviewboard.asterisk.org/r/4193
ASTERISK-24533 #close
Reported by: xrobau
Tested by: xrobau
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a CANCEL has been received.
Due to the serialized architecture of chan_pjsip there exists a race condition where a CANCEL may
be received and processed before responses (such as 180 Ringing, 183 Session Progress, and 200 OK)
are sent. Since the session is in an unexpected state PJSIP will assert when this is attempted.
This change makes it so that these responses are not sent on disconnected sessions.
ASTERISK-24471 #close
Reported by: yaron nahum
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Accessing members of struct ast_str outside of the string manipulation API
routines is invalid since struct ast_str is supposed to be treated as
opaque.
Review: https://reviewboard.asterisk.org/r/4194/
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If transmit_refer is called when p->refer is already allocated,
it leaks the previous allocation. Updated code to always free
previous allocation during a new allocation. Also instead of
checking if we have a previous allocation, always create a
clean record.
ASTERISK-15242 #close
Reported by: David Woolley
Review: https://reviewboard.asterisk.org/r/4160/
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The AOR value may contain the name of an AOR or a full SIP URI.
Checking if the AOR exists can't be done as a result of this.
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and it does not exist.
ASTERISK-24499 #close
Reported by: Rusty Newton
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For chan_motif the direct return value of the underlying config options framework
was passed back. This can relay various states which the module loader would not
interpet as success. It has been changed so only on errors will it report back
an error.
For chan_pjsip the code implemented a dummy reload function which always
returned an error. This has been removed as all configuration is held within
res_pjsip instead.
ASTERISK-23651 #close
Reported by: Rusty Newton
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Configuration of DTLS in the general section will be applied to any users
or peers. If configuration exists at their level it overrides the general
section values.
ASTERISK-24128 #close
Reported by: Michael K.
patches:
dtls_default_settings.patch submitted by Michael K. (license 6621)
Review: https://reviewboard.asterisk.org/r/3867/
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In r227276, a while loop was turned into a for loop. Unfortunately, a portion
of the while loop was left in the code such that, when a static gateway is
encountered in the list of MGCP gateways, the next gateway would be skipped.
At best, we would simply flip past a gateway; at worst, this could lead to a
crash.
ASTERISK-24500 #close
Reported by: Xavier Hienne
patches:
chan_mgcp.patch uploaded by Xavier Hienne (License 6657)
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Fix a bunch of calls to get_active_pvt
where the reference is never released.
ASTERISK-24504 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4152/
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This change adds an option, moh_passthrough, that when enabled will pass
hold and unhold requests through using a SIP re-invite. When placing on
hold a re-invite with sendonly will be sent and when taking off hold a
re-invite with sendrecv will be sent. This allows remote servers to handle
the musiconhold instead of the local Asterisk instance being responsible.
Review: https://reviewboard.asterisk.org/r/4103/
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When r426594 was made, it did not take into account a unit test that verified
that the function properly populated the unsupported buffer. The function
would previously memset the buffer if it detected it had any contents; since
this function can now be called iteratively on successive headers, the unit
tests would now fail. This patch updates the unit tests to reset the buffer
themselves between successive calls, and updates the documentation of the
function to note that this is now required.
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ASTERISK-24304 #close
Reported by: dhanapathy sathya
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This patch adds support for 414, 493, 479, and a stray 400 response in REGISTER
response handling. This helps interoperability in a number of scenarios.
Review: https://reviewboard.asterisk.org/r/3437
patches:
rb3437.patch uploaded by oej (License 5267)
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A SIP request may contain multiple Supported: and Required: headers. Currently,
chan_sip only parses the first Supported/Required header it finds. This patch
adds support for multiple Supported/Required headers for INVITE requests.
Review: https://reviewboard.asterisk.org/r/2478
ASTERISK-21721 #close
Reported by: Olle Johansson
patches:
rb2478.patch uploaded by oej (License 5267)
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A left over from the formats conversion (Corey Farrell).
ASTERISK-24458 #close
Review: https://reviewboard.asterisk.org/r/4117/
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The outboundproxy setting is currently ignored when sending OPTIONS requests
as a result of the qualify setting. This means that if an Asterisk server is
unable to send the packet directly to a peer, it is unable to qualify any
non-inbound registered peer (e.g. a peer SIP Trunk).
This patch grabs the outboundproxy information for a peer when a qualify
attempt is being constructed and, if it finds the information, uses it
when sending the OPTIONS request.
Review: https://reviewboard.asterisk.org/r/3948
ASTERISK-24063 #close
Reported by: Damian Ivereigh
patches:
outboundproxy-dai.patch uploaded by Damian Ivereigh (License 6632)
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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/
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multiple lines on unistim phones. There is regression was introduced in r391379.
Reported by: Rustam Khankishyiev
(closes issue ASTERISK-23846)
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After a reINVITE glare situation, Asterisk would re-send the reINVITE
even though the call had been hung up in the mean time. This patch
unschedules the reinvite when handling the BYE.
ASTERISK-22791 #close
Reported by: Paolo Compagnini
Tested by: Paolo Compagnini
Review: https://reviewboard.asterisk.org/r/4056/
(testcase is in review r4055)
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If a device re-INVITEs at the same time as the dialog is hung up, and
if then the ACK to the re-INVITE never reaches Asterisk, chan_sip would
fail to destroy the dialog after a while. This resulted in (most
prominently) file handle leaks.
(Patch reindented by me.)
ASTERISK-20784 #close
ASTERISK-15879 #close
Reported by: Torrey Searle, Nitesh Bansal
Patches:
reinvite_ack_timeout.patch uploaded by Torrey Searle (License #5334)
patch_asterisk_20784.txt uploaded by Nitesh Bansal (License #6418)
Reviewboard: https://reviewboard.asterisk.org/r/4052/
(testcase can be found at r4051)
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Calling PJSIP_MEDIA_OFFER on a non-PJSIP channel is hazardous to your health.
It will treat the channels as a PJSIP channel, eventually hitting an ao2 error,
FRACKing on assertion error, and quite likely crashing.
This patch adds checks to the read/write callbacks that ensure that the channel
technology is of type 'PJSIP' before attempting to operate on the channel.
#SIPit31
ASTERISK-24382 #close
Reported by: Matt Jordan
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This fix applies to 13 and trunk.
ASTERISK-24384 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4043/
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ASTERISK-24384 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4043/
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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/
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ASTERISK-24222 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4017/
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ASTERISK-22945 #related
Reported by: ibercom
Patches:
asterisk11-chan_sip-simplifies.patch uploaded by ibercom (License #6599)
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The peer is referenced at the end of sip_poke_peer, it should not get
an extra ref before the call to sip_poke_peer. This fixes a memory
leak.
ASTERISK-22945 #close
Reported by: ibercom
Tested by: Yuriy Gorlichenko
Patches:
asterisk11.patch uploaded by ibercom (License #6599)
Review: https://reviewboard.asterisk.org/r/4031/
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