Commit b333ee3b introduced a fix to chan_pjsip that addressed RTP issues with
blind transfers and some SBCs. Unfortunately, the fix broke some WebRTC
clients that are sensitive to SSRC changes and non-monotonic timestamps so
the fix is now disabled for endpoints with the "bundle" parameter set to true.
Resolves: #1535
This change moves the PJSIP module from the threadpool API
to the taskpool API. PJSIP-specific implementations for
task usage have been removed and replaced with calls to
the optimized taskpool implementations instead. The need
for a pool of serializers has also been removed as
taskpool inherently provides this. The default settings
have also been changed to be more realistic for common
usage.
UpgradeNote: The threadpool_* options in pjsip.conf have now
been deprecated though they continue to be read and used.
They have been replaced with taskpool options that give greater
control over the underlying taskpool used for PJSIP. An alembic
upgrade script has been added to add these options to realtime
as well.
Although the ISDN/Q.850/Q.931 hangup cause code is already part of the ARI
and AMI hangup and channel destroyed events, it can be helpful to know what
the actual channel technology code was if the call was unsuccessful.
For PJSIP, it's the SIP response code.
* A new "tech_hangupcause" field was added to the ast_channel structure along
with ast_channel_tech_hangupcause() and ast_channel_tech_hangupcause_set()
functions. It should only be set for off-nominal terminations.
* chan_pjsip was modified to set the tech hangup cause in the
chan_pjsip_hangup() and chan_pjsip_session_end() functions. This is a bit
tricky because these two functions aren't always called in the same order.
The channel that hangs up first will get chan_pjsip_session_end() called
first which will trigger the core to call chan_pjsip_hangup() on itself,
then call chan_pjsip_hangup() on the other channel. The other channel's
chan_pjsip_session_end() function will get called last. Unfortunately,
the other channel's HangupRequest events are sent before chan_pjsip has had a
chance to set the tech hangupcause code so the HangupRequest events for that
channel won't have the cause code set. The ChannelDestroyed and Hangup
events however will have the code set for both channels.
* A new "tech_cause" field was added to the ast_channel_snapshot_hangup
structure. This is a public structure so a bit of refactoring was needed to
preserve ABI compatibility.
* The ARI ChannelHangupRequest and ChannelDestroyed events were modified to
include the "tech_cause" parameter in the JSON for off-nominal terminations.
The parameter is suppressed for nominal termination.
* The AMI SoftHangupRequest, HangupRequest and Hangup events were modified to
include the "TechCause" parameter for off-nominal terminations. Like their ARI
counterparts, the parameter is suppressed for nominal termination.
DeveloperNote: A "tech_cause" parameter has been added to the
ChannelHangupRequest and ChannelDestroyed ARI event messages and a "TechCause"
parameter has been added to the HangupRequest, SoftHangupRequest and Hangup
AMI event messages. For chan_pjsip, these will be set to the last SIP
response status code for off-nominally terminated calls. The parameter is
suppressed for nominal termination.
It is customary to allow # to terminate digit collection immediately
when there would normally be a timeout. However, currently, users are
forced to wait for the timeout to expire when dialing numbers that
are prefixes of other valid matches, and there is no way to end the
timeout early. Customarily, # terminates the timeout, but at the moment,
this is just rejected unless there happens to be a matching extension
ending in #.
Allow # to terminate the timeout in cases where there is no dialplan
match. This ensures that the dialplan is always respected, but if a
valid extension has been dialed that happens to prefix other valid
matches, # can be used to dial it immediately.
Resolves: #1510
If Last Number Redial is used to redial, ensure that we do not wait
for further digits. This was possible if the number that was last
dialed is a prefix of another possible dialplan match. Since all we
did is copy the number into the extension buffer, if other matches
are now possible, there would thus be a timeout before the call went
through. We now complete redialed calls immediaetly in all cases.
Resolves: #1483
Add a dialplan function that can be used to get/set properties of
DAHDI channels (as opposed to Asterisk channels). This exposes
properties that were not previously available, allowing for certain
operations to now be performed in the dialplan.
Resolves: #1455
UserNote: The DAHDI_CHANNEL function allows for getting/setting
certain properties about DAHDI channels from the dialplan.
* Fixed an issue in webchan_write() where we weren't detecting equivalent
codecs properly.
* Added the "p" dialstring option that puts the channel driver in
"passthrough" mode where it will not attempt to re-frame or re-time
media coming in over the websocket from the remote app. This can be used
for any codec but MUST be used for codecs that use packet headers or whose
data stream can't be broken up on arbitrary byte boundaries. In this case,
the remote app is fully responsible for correctly framing and timing media
sent to Asterisk and the MEDIA text commands that could be sent over the
websocket are disabled. Currently, passthrough mode is automatically set
for the opus, speex and g729 codecs.
* Now calling ast_set_read_format() after ast_channel_set_rawreadformat() to
ensure proper translation paths are set up when switching between native
frames and slin silence frames. This fixes an issue with codec errors
when transcode_via_sln=yes.
Resolves: #1462
If Caller ID is disabled for an FXS port, then we should not send any
Caller ID spill on the line, as we have no Caller ID information that
we can/should be sending.
Resolves: #1394
(cherry picked from commit cc9d192114)
It is possible to modify the dialmode setting in the chan_dahdi/sig_analog
private using the CHANNEL function, to modify it during calls. However,
it was not being reset between calls, meaning that if, for example, tone
dialing was disabled, it would never work again unless explicitly enabled.
This fixes the setting by pairing it with a "perm" version of the setting,
as a few other features have, so that it can be reset to the permanent
setting between calls. The documentation is also clarified to explain
the interaction of this setting and the digitdetect setting more clearly.
Resolves: #1378
(cherry picked from commit 9b5ffe513d)
* Added a new option to the WebSocket dial string to capture the additional
URI parameters.
* Added a new API ast_uri_verify_encoded() that verifies that a string
either doesn't need URI encoding or that it has already been encoded.
* Added a new API ast_websocket_client_add_uri_params() to add the params
to the client websocket session.
* Added XML documentation that will show up with `core show application Dial`
that shows how to use it.
Resolves: #1352
UserNote: A new WebSocket channel driver option `v` has been added to the
Dial application that allows you to specify additional URI parameters on
outgoing connections. Run `core show application Dial` from the Asterisk CLI
to see how to use it.
(cherry picked from commit 15371efeab)
ast_websocket_read() receives data into a fixed 64K buffer then continually
reallocates a final buffer that, after all continuation frames have been
received, is the exact length of the data received and returns that to the
caller. process_text_message() in chan_websocket was attempting to set a
NULL terminator on the received payload assuming the payload buffer it
received was the large 64K buffer. The assumption was incorrect so when it
tried to set a NULL terminator on the payload, it could, depending on the
state of the heap at the time, cause heap corruption.
process_text_message() now allocates its own payload_len + 1 sized buffer,
copies the payload received from ast_websocket_read() into it then NULL
terminates it prevent the possibility of the overrun and corruption.
Resolves: #1384
(cherry picked from commit bbb69b115d)
Previously, we were only using # (ST) as a terminator, and not handling
A (STP), B (ST2P), or C (ST3P), which erroneously led to it being
treated as part of the dialed number. Parse any of these as the start
digit.
Resolves: #1301
(cherry picked from commit f5d2ea75c3)
In the WebSocket channel driver, the FLUSH_MEDIA command clears all frames from
the queue but does not reset the frame_queue_length counter.
As a result, the driver incorrectly thinks the queue is full after flushing,
which prevents new multimedia frames from being sent, especially after multiple
flush commands.
This fix sets frame_queue_length to 0 after flushing, ensuring the queue state
is consistent with its actual content.
Fixes: #1304
(cherry picked from commit f03421b03b)
When the RTP media source changes, such as after a blind transfer, the new source introduces a discontinuous timestamp. According to RFC 3550, Section 5.1, an RTP stream's timestamp for a given SSRC must increment monotonically and linearly.
To comply with the standard and avoid a large timestamp jump on the existing SSRC, a new SSRC is generated for the new media stream.
This change resolves known interoperability issues with certain SBCs (like Sonus/Ribbon) that stop forwarding media when they detect such a timestamp violation. This code uses the existing implementation from chan_sip.
Resolves: #927
(cherry picked from commit b333ee3be7)
* Created chan_websocket which can exchange media over both inbound and
outbound websockets which the driver will frame and time.
See http://s.asterisk.net/mow for more information.
* res_http_websocket: Made defines for max message size public and converted
a few nuisance verbose messages to debugs.
* main/channel.c: Changed an obsolete nuisance error to a debug.
* ARI channels: Updated externalMedia to include chan_websocket as a supported
transport.
UserNote: A new channel driver "chan_websocket" is now available. It can
exchange media over both inbound and outbound websockets and will both frame
and re-time the media it receives.
See http://s.asterisk.net/mow for more information.
UserNote: The ARI channels/externalMedia API now includes support for the
WebSocket transport provided by chan_websocket.
(cherry picked from commit 733196abf9)
When a call is transfered via DTMF feature code, the Transfer Target and
Transferer are bridged immediately. This opens the possibilty of a race
condition between the creation of an INVITE and the bridge induced colp
update that can result in the set caller ID being over-written with the
transferer's default info.
Fixes: #1234
Adds support for Call Waiting Deluxe options to enhance
the current call waiting feature.
As part of this change, a mechanism is also added that
allows a channel driver to queue an audio file for Dial()
to play, which is necessary for the announcement function.
ASTERISK-30373 #close
Resolves: #271
UserNote: Call Waiting Deluxe can now be enabled for FXS channels
by enabling its corresponding option.
* Update Dial() documentation for IAX2 to include syntax for RSA
public key names.
* Add additional details to a couple warnings to provide more context
when an undecodable frame is received.
Resolves: #1206
(cherry picked from commit b3e3a0962e)
Updated the AudioSocket protocol to allow sending DTMF frames.
AST_FRAME_DTMF frames are now forwarded to the server, in addition to
AST_FRAME_AUDIO frames. A new payload type AST_AUDIOSOCKET_KIND_DTMF
with value 0x03 was added to the protocol. The payload is a 1-byte
ascii representing the DTMF digit (0-9,*,#...).
UserNote: The AudioSocket protocol now forwards DTMF frames with
payload type 0x03. The payload is a 1-byte ascii representing the DTMF
digit (0-9,*,#...).
(cherry picked from commit c2444b188a)
- Correct wait timeout logic in the dialplan application.
- Include server address in log messages for better traceability.
- Allow dialplan app to exit gracefully on hangup messages and socket closure.
- Optimize I/O by reducing redundant read()/write() operations.
Co-authored-by: Florent CHAUVEAU <florentch@pm.me>
(cherry picked from commit a6ef41e4ca)
1. When one channel is placed on hold, the device state is set to ONHOLD
without checking other channels states.
In case of AST_CONTROL_HOLD set the device state as AST_DEVICE_UNKNOWN
to calculate aggregate device state of all active channels.
2. The current implementation incorrectly classifies channels in use.
The only channels that has the states: UP, RING and BUSY are considered as "in use".
A channel should be considered "in use" if its state is anything other than
DOWN or RESERVED.
3. Currently, if the number of channels "in use" is greater than device_state_busy_at,
the system does not set the state to BUSY. Instead, it incorrectly assigns an aggregate
device state.
The endpoint device state should be BUSY if the number of channels "in use" is greater
than or equal to device_state_busy_at.
Fixes: #1181
(cherry picked from commit b82d9b0d5b)
Found via `codespell -q 3 -S "./CREDITS,*.po" -L abd,asent,atleast,cachable,childrens,contentn,crypted,dne,durationm,enew,exten,inout,leapyear,mye,nd,oclock,offsetp,ot,parm,parms,preceeding,pris,ptd,requestor,re-use,re-used,re-uses,ser,siz,slanguage,slin,thirdparty,varn,varns,ues`
(cherry picked from commit 32cdce18c2)
Introduce a ChannelTransfer event and the ability to notify progress to
ARI. Implement emitting this event from the PJSIP channel instead of
handling the transfer in Asterisk when configured.
Introduce a dialplan function to the PJSIP channel to switch between the
"core" and "ari-only" behavior.
UserNote: Call transfers on the PJSIP channel can now be controlled by
ARI. This can be enabled by using the PJSIP_TRANSFER_HANDLING(ari-only)
dialplan function.
(cherry picked from commit a0d0c47d06)
Also updates the 'since' of applications/functions that existed before
XML documentation was introduced (1.6.2.0).
(cherry picked from commit 67e89b3e77)
* Do a git blame on the embedded XML application or function element.
* From the commit hash, grab the summary line.
* Do a git log --grep <summary> to find the cherry-pick commits in all
branches that match.
* Do a git patch-id to ensure the commits are all related and didn't get
a false match on the summary.
* Do a git tag --contains <commit> to find the tags that contain each
commit.
* Weed out all tags not ..0.
* Sort and discard any .0.0 and following tags where the commit
appeared in an earlier branch.
* The result is a single tag for each branch where the application or function
was defined.
The applications and functions defined in the following files were done by
hand because the XML was extracted from the C source file relatively recently.
* channels/pjsip/dialplan_functions_doc.xml
* main/logger_doc.xml
* main/manager_doc.xml
* res/res_geolocation/geoloc_doc.xml
* res/res_stir_shaken/stir_shaken_doc.xml
(cherry picked from commit f6a193e87e)
* Do a git blame on the embedded XML managerEvent elements.
* From the commit hash, grab the summary line.
* Do a git log --grep <summary> to find the cherry-pick commits in all
branches that match.
* Do a git patch-id to ensure the commits are all related and didn't get
a false match on the summary.
* Do a git tag --contains <commit> to find the tags that contain each
commit.
* Weed out all tags not ..0.
* Sort and discard any .0.0 and following tags where the commit
appeared in an earlier branch.
* The result is a single tag for each branch where the application or function
was defined.
The events defined in res/res_pjsip/pjsip_manager.xml were done by hand
because the XML was extracted from the C source file relatively recently.
Two bugs were fixed along the way...
* The get_documentation awk script was exiting after it processed the first
DOCUMENTATION block it found in a file. We have at least 1 source file
with multiple DOCUMENTATION blocks so only the first one in them was being
processed. The awk script was changed to continue searching rather
than exiting after the first block.
* Fixing the awk script revealed an issue in logger.c where the third
DOCUMENTATION block contained a XML fragment that consisted only of
a managerEventInstance element that wasn't wrapped in a managerEvent
element. Since logger_doc.xml already existed, the remaining fragments
in logger.c were moved to it and properly organized.
(cherry picked from commit 936e88512e)
Most of the configObjects and configOptions that are implemented with
ACO or Sorcery now have `<since>/<version>` elements added. There are
probably some that the script I used didn't catch. The version tags were
determined by the following...
* Do a git blame on the API call that created the object or option.
* From the commit hash, grab the summary line.
* Do a `git log --grep <summary>` to find the cherry-pick commits in all
branches that match.
* Do a `git patch-id` to ensure the commits are all related and didn't get
a false match on the summary.
* Do a `git tag --contains <commit>` to find the tags that contain each
commit.
* Weed out all tags not <major>.<minor>.0.
* Sort and discard any <major>.0.0 and following tags where the commit
appeared in an earlier branch.
* The result is a single tag for each branch where the API was last touched.
configObjects and configOptions elements implemented with the base
ast_config APIs were just not possible to find due to the non-deterministic
way they are accessed.
Also note that if the API call was on modified after it was added, the
version will be the one it was last modified in.
Final note: The configObject and configOption elements were introduced in
12.0.0 so options created before then may not have any XML documentation.
(cherry picked from commit 772221c82a)
Currently, when receiving an unauthenticated call, we keep track
of the negotiated format in the chosenformat, which allows us
to later create the channel using the right format. However,
this was not done for authenticated calls. This meant that in
certain circumstances, if we had not yet received a voice frame
from the peer, only certain other types of frames (e.g. text),
there were no variables containing the appropriate frame.
This led to problems in the jitterbuffer callback where we
unnecessarily bailed out of retrieving a frame from the jitterbuffer.
This was logic intentionally added in commit 73103bdcd5
in response to an earlier regression, and while this prevents
crashes, it also backlogs legitimate frames unnecessarily.
The abort logic was initially added because at this point in the
code, we did not have the negotiated format available to us.
However, it should always be available to us as a last resort
in chosenformat, so we now pull it from there if needed. This
allows us to process frames the jitterbuffer even if voicefmt
and peerfmt aren't set and still avoid the crash. The failsafe
logic is retained, but now it shouldn't be triggered anymore.
Resolves: #1054
(cherry picked from commit 1b0cd8b10d)
This adds the Last Number Redial feature to
simple switch.
UserNote: Users can now redial the last number
called if the lastnumredial setting is set to yes.
Resolves: #437
(cherry picked from commit abfa8206f3)
* channels/pjsip/dialplan_functions_doc.xml: Added xmlns:xi to docs element.
* main/bucket.c: Removed XML completely since the "bucket" and "file" objects
are internal only with no config file.
* main/named_acl.c: Fixed the configFile element name. It was "named_acl.conf"
and should have been "acl.conf"
* res/res_geolocation/geoloc_doc.xml: Added xmlns:xi to docs element.
* res/res_http_media_cache.c: Fixed the configFile element name. It was
"http_media_cache.conf" and should have been "res_http_media_cache.conf".
(cherry picked from commit 15ed357747)
Previously, when AST_CONTROL_RINGING was received by
a DAHDI device, it would set its channel state to
AST_STATE_RINGING. However, an analysis of the codebase
and other channel drivers reveals RINGING corresponds to
physical power ringing, whereas AST_STATE_RING should be
used for audible ringback on the channel. This also ensures
the correct device state is returned by the channel state
to device state conversion.
Since there seems to be confusion in various places regarding
AST_STATE_RING vs. AST_STATE_RINGING, some documentation has
been added or corrected to clarify the actual purposes of these
two channel states, and the associated device state mapping.
An edge case that prompted this fix, but isn't explicitly
addressed here, is that of an incoming call to an FXO port.
The channel state will be "Ring", which maps to a device state
of "In Use", not "Ringing" as would be more intuitive. However,
this is semantic, since technically, Asterisk is treating this
the same as any other incoming call, and so "Ring" is the
semantic state (put another way, Asterisk isn't ringing anything,
like in the cases where channels are in the "Ringing" state).
Since FXO ports don't currently support Call Waiting, a suitable
workaround for the above would be to ignore the device state and
instead check the channel state (e.g. IMPORT(DAHDI/1-1,CHANNEL(state)))
since it will be Ring if the FXO port is idle (but a call is ringing
on it) and Up if the FXO port is actually in use. (In both cases,
the device state would misleadingly be "In Use".)
Resolves: #1029
(cherry picked from commit afd15274e7)
Add a log message for a path that currently silently drops IAX2
frames without indicating that anything is wrong.
(cherry picked from commit be2bea854e)
Currently, when a chan_pjsip channel receives a VIDUPDATE indication,
an RTP VIDUPDATE frame is only queued on a H.264 stream if WebRTC is
enabled on that endpoint. This restriction does not really make sense.
Now, a VIDUPDATE RTP frame is written even if WebRTC is not enabled (as
is the case with VP8, VP9, and H.265 streams).
Resolves: #1013
(cherry picked from commit 45c5d39c58)
Commit 466eb4a52b introduced a regression
which completely broke Feature Group D and E911 signaling, by removing
the call to analog_my_getsigstr, which affected multiple switch cases.
Restore the original behavior for all protocols except Feature Group C
CAMA (MF), which is all that patch was attempting to target.
Resolves: #993
(cherry picked from commit 4f0a506c67)
In some circumstances, it is possible for the do_monitor thread to
erroneously think that a line is on-hook and send an MWI FSK spill
to it when the line is really off-hook and no MWI should be sent.
Commit 0a8b3d3467 previously fixed this
issue in a more readily encountered scenario, but it has still been
possible for MWI to be sent when it shouldn't be. To robustly fix
this issue, query DAHDI for the hook status to ensure we don't send
MWI on a line that is actually still off hook.
Resolves: #928
(cherry picked from commit 4635a5ff4d)
Add dialplan application PJSIPNOTIFY to send either pre-configured
NOTIFY messages from pjsip_notify.conf or with headers defined in
dialplan.
Also adds the ability to send pre-configured NOTIFY commands to a
channel via the CLI.
Resolves: #799
UserNote: A new dialplan application PJSIPNotify is now available
which can send SIP NOTIFY requests from the dialplan.
The pjsip send notify CLI command has also been enhanced to allow
sending NOTIFY messages to a specific channel. Syntax:
pjsip send notify <option> channel <channel>
This patch introduces a new identifier for channels: tenantid. It's
a stringfield on the channel that can be used for general purposes. It
will be inherited by other channels the same way that linkedid is.
You can set tenantid in a few ways. The first is to set it in the
dialplan with the Set and CHANNEL functions:
exten => example,1,Set(CHANNEL(tenantid)=My tenant ID)
It can also be accessed via CHANNEL:
exten => example,2,NoOp(CHANNEL(tenantid))
Another method is to use the new tenantid option for pjsip endpoints in
pjsip.conf:
[my_endpoint]
type=endpoint
tenantid=My tenant ID
This is considered the best approach since you will be able to see the
tenant ID as early as the Newchannel event.
It can also be set using set_var in pjsip.conf on the endpoint like
setting other channel variable:
set_var=CHANNEL(tenantid)=My tenant ID
Note that set_var will not show tenant ID on the Newchannel event,
however.
Tenant ID has also been added to CDR. It's read-only and can be accessed
via CDR(tenantid). You can also get the tenant ID of the last channel
communicated with via CDR(peertenantid).
Tenant ID will also show up in CEL records if it has been set, and the
version number has been bumped accordingly.
Fixes: #740
UserNote: tenantid has been added to channels. It can be read in
dialplan via CHANNEL(tenantid), and it can be set using
Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is recommended to
use the new tenantid option for pjsip endpoints (e.g., tenantid=My
tenant ID) so that it will show up in Newchannel events. You can set it
like any other channel variable using set_var in pjsip.conf as well, but
note that this will NOT show up in Newchannel events. Tenant ID is also
available in CDR and can be accessed with CDR(tenantid). The peer tenant
ID can also be accessed with CDR(peertenantid). CEL includes tenant ID
as well if it has been set.
UpgradeNote: A new versioned struct (ast_channel_initializers) has been
added that gets passed to __ast_channel_alloc_ap. The new function
ast_channel_alloc_with_initializers should be used when creating
channels that require the use of this struct. Currently the only value
in the struct is for tenantid, but now more fields can be added to the
struct as necessary rather than the __ast_channel_alloc_ap function. A
new option (tenantid) has been added to endpoints in pjsip.conf as well.
CEL has had its version bumped to include tenant ID.
Various SIP headers permit a URI to be prefaced with a `display-name`
production that can include characters (like commas and parentheses)
that are problematic for Asterisk's dialplan parser and, specifically
in the case of this patch, the PJSIP_PARSE_URI function.
This patch introduces a new function - `PJSIP_PARSE_URI_FROM` - that
behaves identically to `PJSIP_PARSE_URI` except that the first
argument is now a variable name and not a literal URI.
Fixes#756
Commit f2f397c1a8 previously
made it possible to send Caller ID parameters to FXS stations
which, prior to that, could not be sent.
This change is complementary in that we now handle receiving
all these parameters on FXO lines and provide these up to
the dialplan, via chan_dahdi. In particular:
* If a redirecting reason is provided, the channel's redirecting
reason is set. No redirecting number is set, since there is
no parameter for this in the Caller ID protocol, but the reason
can be checked to determine if and why a call was forwarded.
* If the Call Qualifier parameter is received, the Call Qualifier
variable is set.
* Some comments have been added to explain why some of the code
is the way it is, to assist other people looking at it.
With this change, Asterisk's Caller ID implementation is now
reasonably complete for both FXS and FXO operation.
Resolves: #681
* Add an AMI action to correspond to the "dahdi show status"
command, allowing span information to be retrieved via AMI.
* Show span number and sig type in "dahdi show channels".
Resolves: #673
There were a few references in the embedded documentation XML
where the case didn't match or where the referenced app or function
simply didn't exist any more. These were causing 404 responses
in docs.asterisk.org.
Commit 729cb1d390 added logic to retry
opening DAHDI channels on "dahdi restart" if they failed initially,
up to 1,000 times in a loop, to address cases where the channel was
still in use. However, this retry loop does not use the actual error,
which means chan_dahdi will also retry opening nonexistent channels
1,000 times per channel, causing a flood of unnecessary warning logs
for an operation that will never succeed, with tens or hundreds of
thousands of open attempts being made.
The original patch would have been more targeted if it only retried
on the specific relevant error (likely EBUSY, although it's hard to
say since the original issue is no longer available).
To avoid the problem above while avoiding the possibility of breakage,
this skips the retry logic if the error is ENXIO (No such device or
address), since this will never succeed.
Resolves: #669
