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If endpoint ACLs were specified, they were not being freed
when endpoints were destroyed. On systems with realtime endpoints, this
could add up quickly since each DB lookup would allocate the ACL without
freeing it.
ASTERISK-26731 #close
Reported by Ustinov Artem
Change-Id: Ie1f8bf5b7a0de628c975beba01e69c56893331ad
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Change-Id: Ia2a6158e5fdf311bc2a1c0c43417978de504b1f1
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When channel format changes occurred as a result of an RTP
re-negotiation the bridge was not informed this had happened.
As a result the bridge technology was not re-evaluated and the
channel may have been in a bridge technology that was incompatible
with its formats. The bridge is now unbridged and the technology
re-evaluated when this occurs.
The chan_pjsip module also allowed asymmetric codecs for sending
and receiving. This did not work with all devices and caused one
way audio problems. The default has been changed to NOT do this
but to match the sending codec to the receiving codec. For users
who want asymmetric codecs an option has been added, asymmetric_rtp_codec,
which will return chan_pjsip to the previous behavior.
The codecs returned by the chan_pjsip module when queried by
the bridge_native_rtp module were also not reflective of the
actual negotiated codecs. The nativeformats are now returned as
they reflect the actual negotiated codecs.
ASTERISK-26423 #close
Change-Id: I6ec88c6e3912f52c334f1a26983ccb8f267020dc
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This patch add config to pjsip by endpoint.
;preferred_codec_only=yes
; Respond to a SIP invite with the single most preferred codec
; rather than advertising all joint codec capabilities. This
; limits the other side's codec choice to exactly what we prefer.
ASTERISK-26317 #close
Reported by: AaronAn
Tested by: AaronAn
Change-Id: Iad04dc55055403bbf5ec050997aee2dadc4f0762
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Change-Id: Ied0c06043d1dfef8fdc9c9a808cf89b118119838
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A recent change attempted to optimize startup by not updating contact
status. Instead, code responsible for qualifying contacts updates the
status as it becomes known. The code even accounts for contacts/AORs
that are not set to be qualified.
The problem, though, is when there are no contacts associated with an
endpoint. A common case is when an endpoint is set to register its
contacts but has not done so yet. In this case, prior to registration,
the endpoint's device state will appear to be "not in use" and hints
associated with that device will appear to be "idle". In actuality, the
device state and hint should both appear as "unavailable". The reason
for the failure is that the optimization change made all persistent
endpoint states set to "unknown".
The fix here is to change the hard-coded "unknown" to be "offline"
instead. The default state will be offline until the qualifying code
determines that the contact is actually online. This way, if there are
no contacts at all, then the state stays as offline, and device state
and hints appear correctly.
ASTERISK-26269 #close
Reported by nappsoft
Change-Id: Ie99b84169393983453076f5e9c0d35ff313a456a
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The compilation failed for devmode
--enable DONT_OPTIMIZE
--enable BETTER_BACKTRACES
--enable DO_CRASH
--enable TEST_FRAMEWORK
res_pjsip/pjsip_configuration.c: In function dtls_handler:
res_pjsip/pjsip_configuration.c:974:20: error:
back may be used uninitialized in this function [-Werror=maybe-uninitialized]
int size = strlen(front);
^
cc1: all warnings being treated as errors
Change-Id: I7f082ead0312792a577ec7c73015ba64dabca580
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contact_user, when specified on an endpoint, will override the user
portion of the Contact header on outgoing requests.
Change-Id: Icd4ebfda2f2e44d3ac749d0b4066630e988407d4
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* Eliminated RAII_VAR() usage in
ast_sip_persistent_endpoint_update_state().
* Added a missing allocation failure check to
persistent_endpoint_find_or_create().
* Made persistent_endpoint_find_or_create() create the new object without
a lock as it isn't needed.
* Cleaned up some ao2 container allocation idioms.
* Reordered res_pjsip_mwi.c load_module() and unload_module()
Change-Id: If8ce88fbd82a0c72a37a2388f74f77237a6a36a8
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The new endpoint option allows the PJSIP channel driver's fax_detect
endpoint option to timeout on a call after the specified number of
seconds into a call. The new feature is disabled if the timeout is set
to zero. The option is disabled by default.
ASTERISK-26214
Reported by: Richard Mudgett
Change-Id: Id5a87375fb2c4f9dc1d4b44c78ec8735ba65453d
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* Fix some whitespace in various routines.
* Rename i to iter in persistent_endpoint_update_state().
* Fix off-nominal copy/paste message wording in
persistent_endpoint_contact_deleted_observer()
Change-Id: Id8e34f5d09e7eebac3af22501c44c1110a3e29d8
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The ASTERISK-25904 change-id I8fad8aae9305481469c38d2146e1ba3a56d3108f
patch introduced several regressions when the newly created "Updated"
state goes out for each endpoint registration refresh.
1) It restarted any OPTIONS RTT ping cycle.
2) It would interfere with a currently active ping and throw off that
ping's resulting RTT calculation.
3) It cleared the RTT time each time the endpoint was refreshed.
4) The cleared RTT time was sent out as a statsd update each time.
5) It created two AMI events for each update.
* Revert the original patch and reimplement it. Now the current contact
status state is re-sent instead of the state being momentarily toggled
every time the endpoint refreshes its registration. The statsd events are
not created for the re-sent refresh because they are sent after every
OPTIONS ping.
ASTERISK-26160 #close
Reported by: Matt Jordan
Change-Id: Ie072be790fbb2a8f5c1c874266e4143fa31f66d1
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If specified, incoming SUBSCRIBE requests will be searched for the matching
extension in the indicated context. If no "subscribe_context" is specified,
then the "context" setting is used.
ASTERISK-25471 #close
Change-Id: I3fb7a15f5bc154079bd348c08b7ad1cdd2d5e514
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The patch removes updating all Endpoints' status on startup.
Instead, only non-qualified aors with static contact
and non-qualified non-expired contacts are retrieved from the realtime to
update the endpoint status to ONLINE.
The endpoint name was added to the contact object to simply find the endpoint
that created this contact.
The status of endpoints with qualified aors will be updated by 'qualify'
functions.
ASTERISK-26061 #close
Change-Id: Id324c1776fa55d3741e0c5457ecac0304cb1a0df
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There are a lot of verbose messages about Endpoint and Contact status
changes if there are many dynamic endpoints.
The patch sets verbose level 2 for Endpoint status changes
and verbose level 3 for Contact status changes.
ASTERISK-26055 #close
Change-Id: Ie64e261ddbbc41bfff0f0190241152cc123fe6d7
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With the old SIP module we can use IP access controls per peer.
PJSIP module missing this feature.
This patch added next configuration Endpoint options:
"acl" - list of IP ACL section names in acl.conf
"deny" - List of IP addresses to deny access from
"permit" - List of IP addresses to permit access from
"contact_acl" - List of Contact ACL section names in acl.conf
"contact_deny" - List of Contact header addresses to deny
"contact_permit" - List of Contact header addresses to permit
This patch also better logging failed request:
add custom message instead of "No matching endpoint found"
add SIP method to logging
ASTERISK-25900
Change-Id: I456dea3909d929d413864fb347d28578415ebf02
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A feature of chan_sip that service providers relied upon was the ability to
identify by the Authorization username. This is most often used when customers
have a PBX that needs to register rather than identify by IP address. From my
own experiance, this is pretty common with small businesses who otherwise
don't need a static IP.
In this scenario, a register from the customer's PBX may succeed because From
will usually contain the PBXs account id but an INVITE will contain the caller
id. With nothing recognizable in From, the service provider's Asterisk can
never match to an endpoint and the INVITE just stays unauthorized.
The fixes:
A new value "auth_username" has been added to endpoint/identify_by that
will use the username and digest fields in the Authorization header
instead of username and domain in the the From header to match an endpoint,
or the To header to match an aor. This code as added to
res_pjsip_endpoint_identifier_user rather than creating a new module.
Although identify_by was always a comma-separated list, there was only
1 choice so order wasn't preserved. So to keep the order, a vector was added
to the end of ast_sip_endpoint. This is only used by res_pjsip_registrar
to find the aor. The res_pjsip_endpoint_identifier_* modules are called in
globals/endpoint_identifier_order.
Along the way, the logic in res_pjsip_registrar was corrected to match
most-specific to least-specific as res_pjsip_endpoint_identifier_user does.
The order is:
username@domain
username@domain_alias
username
Auth by username does present 1 problem however, the first INVITE won't have
an Authorization header so the distributor, not finding a match on anything,
sends a securty_alert. It still sends a 401 with a challenge so the next
INVITE will have the Authorization header and presumably succeed. As a result
though, that first security alert is actually a false alarm.
To address this, a new feature has been added to pjsip_distributor that keeps
track of unidentified requests and only sends the security alert if a
configurable number of unidentified requests come from the same IP in a
configurable amout of time. Those configuration options have been added to
the global config object. This feature is only used when auth_username
is enabled.
Finally, default_realm was added to the globals object to replace the hard
coded "asterisk" used when an endpoint is not yet identified.
The testsuite tests all pass but new tests are forthcoming for this new
feature.
ASTERISK-25835 #close
Reported-by: Ross Beer
Change-Id: I30ba62d208e6f63439600916fcd1c08a365ed69d
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res_pjsip_mwi was missing the chan_sip "vmexten" functionality which adds
the Message-Account header to the MWI NOTIFY. Also, specifying mailboxes
on endpoints for unsolicited mwi and on aors for subscriptions required
that the admin know in advance which the client wanted. If you specified
mailboxes on the endpoint, subscriptions were rejected even if you also
specified mailboxes on the aor.
Voicemail extension:
* Added a global default_voicemail_extension which defaults to "".
* Added voicemail_extension to both endpoint and aor.
* Added ast_sip_subscription_get_dialog for support.
* Added ast_sip_subscription_get_sip_uri for support.
When an unsolicited NOTIFY is constructed, the From header is parsed, the
voicemail extension from the endpoint is substituted for the user, and the
result placed in the Message-Account field in the body.
When a subscribed NOTIFY is constructed, the subscription dialog local uri
is parsed, the voicemail_extension from the aor (looked up from the
subscription resource name) is substituted for the user, and the result
placed in the Message-Account field in the body.
If no voicemail extension was defined, the Message-Account field is not added
to the NOTIFY body.
mwi_subscribe_replaces_unsolicited:
* Added mwi_subscribe_replaces_unsolicited to endpoint.
The previous behavior was to reject a subscribe if a previous internal
subscription for unsolicited MWI was found for the mailbox. That remains the
default. However, if there are mailboxes also set on the aor and the client
subscribes and mwi_subscribe_replaces_unsolicited is set, the existing internal
subscription is removed and replaced with the external subscription. This
allows an admin to configure mailboxes on both the endpoint and aor and allows
the client to select which to use.
ASTERISK-25865 #close
Reported-by: Ross Beer
Change-Id: Ic15a9415091760539c7134a5ba3dc4a6a1217cea
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Added the ability to show channel statistics to chan_pjsip (cli_functions.c)
Moved the existing 'pjsip show channel(s)' functionality from
pjsip_configuration to cli_functions.c. The stats needed chan_pjsip's
private header so it made sense to move the existing channel commands as well.
Now using stasis_cache_dump to get the channel snapshots rather than retrieving
all endpoints, then getting each one's channel snapshots. Much more efficient.
Change-Id: I03b114522126d27434030b285bf6d531ddd79869
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Configurations like "aors = a, b, c" were either ignoring everything after "a"
or trying to look up " b". Same for mailboxes, ciphers, contacts and a few
others.
To fix, all the strsep(©, ",") calls have been wrapped in ast_strip. To
facilitate this, ast_strip, ast_skip_blanks and ast_skip_nonblanks were
updated to handle null pointers.
In some cases, an ast_strlen_zero() test was added to skip consecutive commas.
There was also an attempt to ast_free an ast_strdupa'd string in
ast_sip_for_each_aor which was causing a SEGV. I removed it.
Although this issue was reported for realtime, the issue was in the res_pjsip
modules so all config mechanisms were affected.
ASTERISK-25829 #close
Reported-by: Mateusz Kowalski
Change-Id: I0b22a2cf22a7c1c50d4ecacbfa540155bec0e7a2
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load_module was just too hairy with every step having to clean up all
previous steps on failure.
Some of the pjproject init calls have now been moved to a separate
load_pjsip function and the unload_pjsip function was enhanced to clean
up everything if an error happened at any stage of the load process.
In the process, a bunch of missing pj_shutdowns, serializer_pool_shutdowns
and ast_threadpool_shutdowns were also corrected.
Change-Id: I5eec711b437c35b56605ed99537ebbb30463b302
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Added new global option (regcontext) to pjsip. When set, Asterisk will
dynamically create and destroy a NoOp priority 1 extension
for a given endpoint who registers or unregisters with us.
ASTERISK-25670 #close
Reported-by: Daniel Journo
Change-Id: Ib1530c5b45340625805c057f8ff1fb240a43ea62
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On a system with multiple ip addresses in the same subnet, if a
transport is bound to a specific ip address and endpoint/media_address
is set, the SIP/SDP will have the correct address in all fields but
the rtp stream MAY still originate from one of the other ip addresses,
most probably the "primary" ip address. This happens because
res_pjsip_sdp_rtp/create_rtp always calls ast_instance_new with
the "all" ip address (0.0.0.0 or ::).
The new option causes res_pjsip_sdp_rtp/create_rtp to call
ast_rtp_instance_new with the endpoint's media_address (if specified)
instead of the "all" address. This causes the packets to originate from
the specified address.
ASTERISK-25632
ASTERISK-25637
Reported-by: Olivier Krief
Reported-by: Dan Journo
Change-Id: I3dfaa079e54ba7fb7c4fd1f5f7bd9509bbf8bd88
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Both transport and endpoint now check for the existence and readability
of tls certificate and key files before passing them on to pjproject.
This will cause the object to not load rather than waiting for pjproject
to discover that there's a problem when a session is attempted.
NOTE: chan_sip also uses ast_rtp_dtls_cfg_parse but it's located
in build_peer which is gigantic and I didn't want to disturb it.
Error messages will emit but it won't interrupt chan_sip loading.
ASTERISK-25618 #close
Change-Id: Ie43f2c1d653ac1fda6a6f6faecb7c2ebadaf47c9
Reported-by: George Joseph
Tested-by: George Joseph
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It will never be perfect or even pretty, mostly because of the differences
between static and dynamic contacts.
Created:
Can't use the contact or contact_status alloc functions
because the objects come and go regardless of the actual state.
Can't use the contact_apply_handler, ast_sip_location_add_contact or
a sorcery created handler because they only get called for dynamic
contacts. Similarly, permanent_uri_handler only gets called for
static contacts.
So, Matt had it right. :) ast_res_pjsip_find_or_create_contact_status is
the only place it can go and not have duplicated code. Both
permanent_uri_handler and contact_apply_handler call find_or_create.
Removed:
Can't use the destructors for the same reason as above. The only
place to put this is in persistent_endpoint_contact_deleted_observer
which I believe is the "correct" place but even that will handle only
dynamic contacts. This doesn't called on shutdown however. There is
no hook to use for static contacts that may be removed because of a
config change while asterisk is in operation.
I moved the cleanup of contact_status from ast_sip_location_delete_contact
to the handler as well.
Status Change and RTT:
Although they worked fine where they were (in update_contact_status) I
moved them to persistent_endpoint_contact_status_observer to make it
more consistent with removed. There was logic there already to detect
a state change.
Finally, fixed a nit in permanent_uri_handler rmudgett reported
eralier.
ASTERISK-25608 #close
Change-Id: I4b56e7dfc3be3baaaf6f1eac5b2068a0b79e357d
Reported-by: George Joseph
Tested-by: George Joseph
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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
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When an endpoint is deleted (such as through an API), the persistent endpoint
currently continues to lurk around. While this isn't harmful from a memory
consumption perspective - as all persistent endpoints are reclaimed on
shutdown - it does cause Stasis endpoint related operations to continue
to believe that the endpoint may or may not exist.
This patch causes the persistent endpoint related to a PJSIP endpoint to be
destroyed if the PJSIP endpoint is deleted.
Change-Id: I85ac707b4d5e6aad882ac275b0c2e2154affa5bb
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Add the ability to filter output from pjsip list and show commands
using the "like" predicate like chan_sip.
For endpoints, aors, auths, registrations, identifyies and transports,
the modification was a simple change of an ast_sorcery_retrieve_by_fields
call to ast_sorcery_retrieve_by_regex. For channels and contacts a
little more work had to be done because neither of those objects are
true sorcery objects. That was just removing the non-matching object
from the final container. Of course, a little extra plumbing in the
common pjsip_cli code was needed to parse the "like" and pass the regex
to the get_container callbacks.
Some of the get_container code in res_pjsip_endpoint_identifier was also
refactored for simplicity.
ASTERISK-25477 #close
Reported by: Bryant Zimmerman
Tested by: George Joseph
Change-Id: I646d9326b778aac26bb3e2bcd7fa1346d24434f1
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In a realtime based system with a limited number of threadpool threads
it is possible for a deadlock to occur. This happens when permanent
endpoint state is updated, which will cause database queries to be done.
These queries may result in URI validation being done which is done
synchronously using a PJSIP thread. If all PJSIP threads are in use
processing traffic they themselves may be blocked waiting to get the
permanent endpoint container lock when identifying an endpoint.
This change moves URI validation to occur at use time instead of
configuration time. While this comes at a cost of not seeing a problem
until you use it it does solve the underlying deadlock problem.
ASTERISK-25486 #close
Change-Id: I2d7d167af987d23b3e8199e4a68f3359eba4c76a
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When an endpoint is backed by a non-static conf file backend (such as
the AstDB or Realtime), the 'auth' object may be returned as being an
empty string. Currently, res_pjsip will interpret that as being a valid
auth object, and will attempt to authenticate inbound requests. This
isn't desired; is an auth value is empty (which the name of an auth
object cannot be), we should instead interpret that as being an invalid
auth object and skip it.
ASTERISK-25339 #close
Change-Id: Ic32b0c6eb5575107d5164a8c40099e687cd722c7
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This change adds support for the 'rtp_timeout' and 'rtp_timeout_hold'
endpoint options. These allow the channel to be hung up if RTP
is not received from the remote endpoint for a specified number of
seconds.
ASTERISK-25259 #close
Change-Id: I3f39daaa7da2596b5022737b77799d16204175b9
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This adds an "rtp_keepalive" option for PJSIP endpoints. Similar to the
chan_sip option, this specifies an interval, in seconds, at which we
will send RTP comfort noise frames. This can be useful for keeping RTP
sessions alive as well as keeping NAT associations alive during lulls.
ASTERISK-25242 #close
Reported by Mark Michelson
Change-Id: I3b9903d99e35fe5d0b53ecc46df82c750776bc8d
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This patch fixes some bad default value handling in the following
settings:
* The 'message_context' and 'accountcode' settings are not mandatory. As
such, we can allow their stringfield values to be empty.
* The 'media_encryption' setting applies a default value of 'none' to
the setting, which it then can't parse or understand. Since the value
is documented to be 'no', this will now apply that as the default
value.
Change-Id: Ib9be7f97a7a5b9bc7aee868edf5acf38774cff83
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Some phones send g.726 audio packed for AAL2, which differs from what is
recommended by RFC 3351. If Asterisk receives audio formatted as such when
negotiating g.726 then it sounds a bit distorted. Added an option to
res_pjsip_endpoint that allows g.726 negotiated audio to be treated as g.726
AAL2 packed.
ASTERISK-25158 #close
Reported by: Steve Pitts
Change-Id: Ie7e21f75493d7fe53e75e12c971e72f5afa33615
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The loop to find the first available contact of an endpoint grabbed
contact from the iterator, then checked for offline state. This
caused the first contact after the state was found to leak a reference.
ASTERISK-25141
Change-Id: Id0f1d87410fc63742db0594eb4b18b36e99aec08
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Add a new ContactStatus AMI event.
Publish the following status/state changes:
Created
Removed
Reachable
Unreachable
Unknown
Contact URI, new status/state, aor and endpoint names, and the
last qualify rtt result are included in the event.
ASTERISK-25114 #close
Change-Id: Id25aae5f7122facba183273efb3e8f36c20fb61e
Reported-by: George Joseph <george.joseph@fairview5.com>
Tested-by: George Joseph <george.joseph@fairview5.com>
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This patch refactors the transaction timeout processing to eliminate
calling the lower level public pjsip functions and reverts to calling
pjsip_endpt_send_request again. This is the result of me noticing
a possible incompatibility with pjproject-2.4 which was causing
contact status flapping.
The original version of this feature used the lower level calls to
get access to the tsx structure in order to cancel the transaction
when our own timer expires. Since we no longer have that access,
if our own timer expires before the pjsip timer, we call the callbacks
and just let the pjsip transaction take it's own course. When the
transaction ends, it discovers the callbacks have already been run
and just cleans itself up.
A few messages in pjsip_configuration were also added/cleaned up.
ASTERISK-25105 #close
Change-Id: I0810f3999cf63f3a72607bbecac36af0a957f33e
Reported-by: George Joseph <george.joseph@fairview5.com>
Tested-by: George Joseph <george.joseph@fairview5.com>
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ast_module_info->self is often needed to register items with the core. Many
modules have ad-hoc code to make this pointer available to auxiliary sources.
This change updates the module build process to make the needed information
available to all sources in a module.
ASTERISK-25056 #close
Reported by: Corey Farrell
Change-Id: I18c8cd58fbcb1b708425f6757becaeca9fa91815
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The "Add qualify_timeout processing and eventing" patch introduced
an issue where contacts that had qualify_frequency set to 0 were
showing Unavailable instead Unknown. This patch checks for
qualify_frequency=0 and create an "Unknown" contact_status
with an RTT = 0.
Previously, the lack of contact_status implied Unknown but since
we're now changing endpoint state based on contact_status, I've
had to add new UNKNOWN status so that changes could trigger the
appropriate contact_status observers.
ASTERISK-24977: #close
Change-Id: Ifcbc01533ce57f0e4e584b89a395326e098b8fe7
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This is the second follow-on to https://reviewboard.asterisk.org/r/4572/ and the
discussion at
http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html
The basic issues are that changes in contact status don't cause events to be
emitted for the associated endpoint. Only dynamic contact add/delete actions
update the endpoint. Also, the qualify timeout is fixed by pjsip at 32 seconds
which is a long time.
This patch makes use of the new transaction timeout feature in r4585 and
provides the following capabilities...
1. A new aor/contact variable 'qualify_timeout' has been added that allows the
user to specify the maximum time in milliseconds to wait for a response to an
OPTIONS message. The default is 3000ms. When the timer expires, the contact is
marked unavailable.
2. Contact status changes are now propagated up to the endpoint as follows...
When any contact is 'Available', the endpoint is marked as 'Reachable'. When
all contacts are 'Unavailable', the endpoint is marked as 'Unreachable'. The
existing endpoint events are generated appropriately.
ASTERISK-24863 #close
Change-Id: Id0ce0528e58014da1324856ea537e7765466044a
Tested-by: Dmitriy Serov
Tested-by: George Joseph <george.joseph@fairview5.com>
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This patch adds support for automatically detecting the type of DTMF that a
PJSIP endpoint supports. When the 'dtmf_mode' endpoint option is set to 'auto',
the channel created for an endpoint will attempt to determine if RFC 4733
DTMF is supported. If so, it will use that DTMF type. If not, the DTMF type
for the channel will be set to inband.
Review: https://reviewboard.asterisk.org/r/4438
ASTERISK-24706 #close
Reported by: yaron nahum
patches:
yaron_patch_3_Feb.diff submitted by yaron nahum (License 6676)
........
Merged revisions 434637 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434638 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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When setting the configuration option 'timers' equal to 'no' the bit flag was
not properly negated. This patch clears all associated flags and only sets the
specified one. pjsip will handle any necessary flag combinations. Also went
ahead and did similar for the '100rel' option.
ASTERISK-24910 #close
Reported by: Ray Crumrine
Review: https://reviewboard.asterisk.org/r/4582/
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messages.
Incoming PJSIP call legs that have not been answered yet send unnecessary
"180 Ringing" or "183 Progress" messages every time a connected line
update happens. If the outgoing channel is also PJSIP then the incoming
channel will always send a "180 Ringing" or "183 Progress" message when
the outgoing channel sends the INVITE.
Consequences of these unnecessary messages:
* The caller can start hearing ringback before the far end even gets the
call.
* Many phones tend to grab the first connected line information and refuse
to update the display if it changes. The first information is not likely
to be correct if the call goes to an endpoint not under the control of the
first Asterisk box.
When connected line first went into Asterisk in v1.8, chan_sip received an
undocumented option "rpid_immediate" that defaults to disabled. When
enabled, the option immediately passes connected line update information
to the caller in "180 Ringing" or "183 Progress" messages as described
above.
* Added "rpid_immediate" option to prevent unnecessary "180 Ringing" or
"183 Progress" messages. The default is "no" to disable sending the
unnecessary messages.
ASTERISK-24781 #close
Reported by: Richard Mudgett
Review: https://reviewboard.asterisk.org/r/4473/
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Also fixed similar problem with AMI action PJSIPShowEndpoints.
ASTERISK-24872 #close
Reported by: Dmitriy Serov
Review: https://reviewboard.asterisk.org/r/4487/
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When a type=global section is not defined in pjsip.conf the global
defaults are not applied. As a result the mandatory Max-Forwards header
is not added to SIP messages for res_pjsip/chan_pjsip.
The handling of pjsip.conf type=global objects has several problems:
1) If the global object is missing the defaults are not applied.
2) If the global object is missing the default_outbound_endpoint's default
value is not returned by ast_sip_global_default_outbound_endpoint().
3) Defines are needed so default values only need to be changed in one
place.
* Added a sorcery instance observer callback to check if there were any
type=global sections loaded. If there were more than one then issue an
error message. If there were none then apply the global defaults.
* Fixed ast_sip_global_default_outbound_endpoint() to return the
documented default when no type=global object is defined.
* Made defines for the global default values.
* Increased the default_useragent[] size because SVN version strings can
get lengthy and 128 characters may not be enough.
* Fixed an off-nominal code path ref leak in global_alloc() if the string
fields fail to initialize.
* Eliminated RAII_VAR in get_global_cfg() and
ast_sip_global_default_outbound_endpoint().
ASTERISK-24807 #close
Reported by: Anatoli
Review: https://reviewboard.asterisk.org/r/4467/
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There have been a couple of times where a crash occurred in the dtls_handler
section of the code for res_pjsip. Unfortunately, in working this issue the
problem was unable to be reproduced. After looking at the backtraces and
through the code the current best guess as to why this happened might be due
to a reentrance problem and the strtok function. So, the current fix is to
convert the strtok function into the reentrant version of the function,
strtok_r.
ASTERISK-24741 #close
Reported by: Zane Conkle
Review: https://reviewboard.asterisk.org/r/4409/
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Due to an inversion error, setting 100rel=no would not actually
change the current value of the setting (which defaulted to "yes").
With this fix, the inversion is corrected.
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