| Age | Commit message (Collapse) | Author |
|---|
|
When 4875e5ac32 was merged, it fixed several issues with a direct media bridge
transitioning to handling a T.38 fax. However, it uncovered a race condition
caused by the bridging core. When a channel involved in a T.38 fax leaves a
bridge, the frame queued by the channel driver that should inform the far side
that it is no longer in a T.38 fax may not make it across the bridge. The
bridging framework is *extremely* aggressive in tearing down the bridge, and
control frames that are currently in flight *may* get dropped.
This patch adds a new module to the bridging framework, bridge_t38. This module
maintains some notion of the T.38 state for the two channels in a bridge. When
the bridge detects that it is being torn down or when one of the two channels
leaves, it informs the respective channel(s) that they should stop faxing. This
ensures that channels switch back to audio if they survive and are ejected out
of a bridge while faxing.
ASTERISK-25582
Change-Id: If5b0bb478eb01c4607c9f4a7fc17c7957d260ea0
|
|
|
|
|
|
|
|
|
|
When 90d9a70789 was merged, it mostly tested dynamic contacts created as
a result of registering a PJSIP endpoint. Contacts generated in this
fashion typically have a long alphanumeric string as their object identifier,
which maps reasonably well for StatsD. Unfortunately, this doesn't work in the
general case. StatsD treats both '.' and ':' characters as special characters.
In particular, having a ':' appear in the middle of a StatsD metric will
result in the metric being rejected.
This causes some obvious issues with SIP URIs.
The StatsD API should not be responsible for escaping the metric name passed
to it. The metric is treated as a single long string, and it would be
challenging to know what to escape in the string passed to the function.
Likewise, we don't want to escape the metric in PJSIP, as that involves
overhead that is wasted when either res_statsd isn't loaded or enabled.
This patch takes an alternative approach. The Contact ID has been changed
to be "aor@@uri_hash" instead of "aor@@uri". This (a) won't contain any of the
aforementioned special characters, (b) can be done on Contact creation,
which has minimal impact on run-time performance, and (c) also conforms to an
earlier commit that changed the ID for dynamic contacts.
The downside of this is that StatsD users will have to map SHA1 hashes back to
the Contacts that are emitting the statistics. To that end, the CLI commands
have been updated to include the first 10 characters of the MD5 hash, which
should be enough to match what is shown in Graphite (or some other StatsD
backend).
ASTERISK-25595 #close
Change-Id: Ic674a3307280365b4a45864a3571c295b48a01e2
Reported-by: Matt Jordan
Tested-by: George Joseph
|
|
|
|
|
|
|
|
additions)"
|
|
|
|
An earlier commit changed the id of dynamic contacts to contain
a hash instead of the uri. This patch updates status change
logging to show the aor/uri instead of the id. This required
adding the aor id to contact and contact_status and adding
uri to contact_status. The aor id gets added to contact and
contact_status in their allocators and the uri gets added to
contact_status in pjsip_options when the contact_status is
created or updated.
ASTERISK-25598 #close
Reported-by: George Joseph
Tested-by: George Joseph
Change-Id: I56cbec1d2ddbe8461367dd8b6da8a6f47f6fe511
|
|
Currently if a channel is transferred out of a bridge, the BRIDGEPEER
variable (also BRIDGEPVTCALLID) remain set even once the channel is
out of the bridge. This patch removes these variables when leaving
the bridge.
ASTERISK-25600 #close
Reported by: Mark Michelson
Change-Id: I753ead2fffbfc65427ed4e9244c7066610e546da
|
|
Change-Id: If83d63cf11cbc6df9b15251848b01feb570ade49
|
|
According to the API doxygen a sched ID of 0 is valid. Unfortunately, 0
was never returned historically and several users incorrectly coded usage
of the returned sched ID assuming that 0 was invalid.
ASTERISK-25476
Change-Id: Ib19c7ebb44ec9fd393ef6646dea806d4f34e3a20
|
|
chan_sip.c:
* Initialize mwi subscription scheduler ids earlier because of ASTOBJ to
ao2 conversion.
* Initialize register scheduler ids earlier because of ASTOBJ to ao2
conversion.
chan_skinny.c:
* Fix more scheduler usage for the valid 0 id value.
ASTERISK-25476
Change-Id: If9f0e5d99638b2f9d102d1ebc9c5a14b2d706e95
|
|
channels/chan_iax2.c:
* Initialize struct chan_iax2_pvt scheduler ids earlier because of
iax2_destroy_helper().
channels/chan_sip.c:
channels/sip/config_parser.c:
* Fix initialization of scheduler id struct members. Some off nominal
paths had 0 as a scheduler id to be destroyed when it was never started.
chan_skinny.c:
* Fix some scheduler id comparisons that excluded the valid 0 id.
channel.c:
* Fix channel initialization of the video stream scheduler id.
pbx_dundi.c:
* Fix channel initialization of the packet retransmission scheduler id.
ASTERISK-25476
Change-Id: I07a3449f728f671d326a22fcbd071f150ba2e8c8
|
|
ASTERISK-25599 #close
Change-Id: I1f88a88c59fb4e1e62bbdbb100c7152d48e73f10
|
|
dns.c and dns_system_resolver.c were spitting out errors for lookup
failures for things like not finding a SRV record even though
there was an A record. Those have been changed to debug messages.
Logging not finding ANY record is left to the higher level caller.
Also, dns_system_resolver was using Windows line endings so I
converted them to Unix style. The actual log changes are on lines
156 and 159.
Change-Id: I65be16ea15304b96f9dcb4d289dbd3e2286fc094
|
|
ASTERISK-25591 #close
Change-Id: I8d3efa0826142ece9cbed2fd0d46f3b607fee6ae
|
|
Add value of pause reason when is paused on CLI command "queue show"
ASTERISK-25581 #close
Report by: Rodrigo Ramírez Norambuena
Change-Id: I887028a40cd97b350da9a3bb2719616b7fec9864
|
|
Change-Id: Iceb3d9bb78140c376174a7bee197dfcf8ef9cda7
|
|
|
|
|
|
The fastagi record-file testsuite test sometimes fails reporting an empty
recorded file. This was happening because Asterisk was sending the agi result
notification prior to actually closing the file and the data, being buffered,
had not been written to the file yet when the test attempts to check the file
size.
This patch makes it so the record file stream is closed prior to sending the
agi result notification.
ASTERISK-25593 #close
Change-Id: I6b2b3be3ae37f7c7b18e672c419a89b3b8513cde
|
|
Several issues are addressed here:
- main() is large, and half of it is only used if we're not rasterisk;
fixed by spliting up the daemon part into a separate function.
- Call ast_term_init from rasterisk as well.
- Remove duplicate code reading/writing asterisk history file.
- Attempt to tackle background color issues and color changes that
occur. Tested by starting asterisk -c until the colors stopped
changing at odd locations.
- Remove unused term_prep() and term_prompt() functions.
ASTERISK-25585 #close
Change-Id: Ib641a0964c59ef9fe6f59efa8ccb481a9580c52f
|
|
|
|
Fixes some minor typos in the CHANGES file, plus an embarrasing typo in
the StatsD API.
Change-Id: I9ca4858c64a4a07d2643b81baa64baebb27a4eb7
|
|
The usage info for 'pjsip send notify' previously referenced the
chan_sip configuration sip_notify.conf. Fix this to reference
the correct configuration pjsip_notify.conf.
ASTERISK-25590 #close
Change-Id: I3898271a8e8a8b1db201741e790ebe2c6bf5cdea
|
|
|
|
|
|
This patch adds a module that emits StatsD statistics about Asterisk
endpoints. This includes:
* A GAUGE statistic for endpoint states, tracking how many endpoints are in
a particular state.
* A GAUGE statistic for each endpoint, counting the number of channels
currently associated with an endpoint.
ASTERISK-25572
Change-Id: If7e1333c5aeda8d136850b30c2101c0ee1c97305
|
|
If the sorcery object type is not found a NULL is returned.
Unfortunately, sorcery_realtime_filter_objectset() will crash after
complaining about not finding the object type and saying to expect errors.
* Use ao2_cleanup() instead of ao2_ref() to prevent the crash.
ASTERISK-25165
Reported by Corey Farrell
Change-Id: Ic3b64453ea3058cb68d5c26d97d4fe7b8eea2e97
|
|
|
|
|
|
This patch adds the ability to send StatsD statistics related to the
state of PJSIP contacts. This includes:
* A GUAGE statistic measuring the count of contacts in a particular state.
This measures how many contacts are reachable, unreachable, etc.
* The RTT time for each contact, if those contacts are qualified. This
provides StatsD engines useful time-based data about each contact.
ASTERISK-25571
Change-Id: Ib8378d73afedfc622be0643b87c542557e0b332c
|
|
This patch adds outbound registration statistics for StatsD. This includes
the following:
* A GUAGE metric for the overall count of outbound registrations.
* A GUAGE metric for each state an outbound registration can be in. As the
outbound registrations change state, the overall count of how many
outbound registrations are in the particular state is changed.
These statistics are particularly useful for systems with a large number of
SIP trunks, and where measuring the change in state of the trunks is useful
for monitoring.
ASTERISK-25571
Change-Id: Iba6ff248f5d1c1e01acbb63e9f0da1901692eb37
|
|
Often, the metric names of statistics we are generating for StatsD have some
dynamic component to them. This can be the name of a particular resource, or
some internal status label in Asterisk. With the current set of functions,
callers of the statsd API must first build the metric name themselves, then
pass this to the API functions. This results in a large amount of boilerplate
code and usage of either fixed length static buffers or dynamic memory
allocation, neither of which is desireable.
This patch adds two new functions to the StatsD API that support a printf
style format specifier for constructing the metric name. A dynamic string,
allocated in threadstorage, is used to build the metric name. This eases
the burden on users of the StatsD API.
Change-Id: If533c72d1afa26d807508ea48b4d8c7b32f414ea
|
|
When a channel is in a direct media bridge, a re-INVITE may arrive that forces
Asterisk to re-negotiate the media to a T.38 fax. When this occurs, the bridge
must change its technology to a simple bridge, and re-INVITE the media back
to Asterisk.
Generally, this logic mostly already exists in Asterisk. However, prior to this
patch, there were a few bugs:
(1) The T.38 framehook currently prevents a channel capable of T.38 faxes from
ever entering into a direct media bridge. This applies even when the only
media being passed over the channel is audio. This patch fixes this bug
by having the framehook specify that it defers caring about any frame type.
This allows the channels to enter into a direct media bridge, which will
be broken when a re-INVITE is received.
(2) When a re-INVITE is received, nothing instructed the bridging layer to
re-inspect the allowed bridging technology. This now occurs when either
a re-INVITE is received from a peer, or when a response is received from
the far end (that is, when the T.38 state changes to either
T38_PEER_REINVITE or T38_LOCAL_REINVITE).
(3) chan_pjsip needs to do a small amount of work to prevent a direct media
bridge from being chosen when a T.38 session is in progress. When a T.38
session supplement has a t38 datastore - which is added when we detect
we should start thinking about T.38 on a channel - we now refuse a native
RTP bridge.
(4) When a BYE request is received, we don't terminate the T.38 session. If
the other side of a T.38 fax survives the hangup (due to the 'g' flag
in Dial, for example), we don't currently re-INVITE the media on the
other channel back to audio. This patch now has res_pjsip_t38 intercept
BYE requests and inform the far side that the T.38 session is terminated.
This naturally causes the correct re-INVITEs to be sent.
ASTERISK-25582
Change-Id: Iabd6aa578e633d16e6b9f342091264e4324a79eb
|
|
context/exten/app"
|
|
|
|
timeouts."
|
|
Because the context, extension, and application are stored in stringfields,
checking for them being NULL doesn't work so well. This patch uses the
appropriate string library call, ast_strlen_zero, to see if there is a value
in the context/exten/app values.
Change-Id: Ie09623bfdf35f5a8d3b23dd596647fe3c97b9a23
|
|
This patch adds some debug statements to res_pjsip_t38. These statements help
to determine which SDP negotiation callbacks are being executed, and, when
a particular callback exits, why a callback may not have applied its logic
to the local or remote SDP.
Change-Id: I61b3fb9183b7ebbb5da8e9f48b59a5d9d7042d77
|
|
|
|
|
|
type load"
|
|
|
|
When Asterisk is configured to use a dynamic sorcery backend (such as
res_sorcery_astdb) with 'registration' objects, it will fail to create the
internal state objects associated with the registration objects on module
load. This is due to nothing actually querying for the specific objects
and calling their sorcery apply handler during module load.
This patch fixes that by calling get_registrations in the sorcery observer's
object_type_loaded handler. Doing this causes the sorcery backends to be
asked for the current state of all registration objects, which causes the
apply handler to be called and the internal run-time state to be created.
ASTERISK-25575 #close
Change-Id: Ie9306e797098c6d4da7bcf4a5434a15891508b23
|
|
Previously, a trancoding module did not have access to the joint but cached
format. Therefore, the module did not have access to the attributes negotiated
via SDP (line fmtp). Now, a translation module receives the joint format.
ASTERISK-25545 #close
Change-Id: Id6878a989b50573298dab115d3371ea369e1a718
|
