The setting 'use_ptime' is supposed to tell Asterisk to honour the ptime
attribute in an offer, preferring it to whatever packetization
preferences have been set internally. Currently, however, something
rather quirky will happen:
(1) The SDP answer will be constructed in create_outgoing_sdp_stream.
This will use the preferences from the endpoint, such that the 200 OK
response will add the packetization preferences from the endpoint, and
not what was offered.
(2) When the 200 response is issued, apply_negotiated_sdp_stream is called.
This will call apply_packetization, which will use the ptime attribute
from the offer internally.
We end up telling the offerer to use the internal ptime attribute, but we end
up using the offered ptime attribute. Hilarity ensues.
This patch modifies the behaviour by calling apply_packetization from
negotiate_incoming_sdp_stream, which is called prior to
create_outgoing_sdp_stream. This causes the format preferences on the
session's media object to be set to the inbound ptime value (if 'use_ptime'
is enabled), such that the construction of the answer gets the right value
immediately.
Review: https://reviewboard.asterisk.org/r/3244/
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It is currently possible for an ast_sip_session to exist without an
associated channel as is the case when a new invite is coming in or
just after a hangup is issued on a chan_pjsip channel. Part of the
attended transfer code assumed the channel would be non-NULL and used
it as such causing a crash. This bug was exposed thanks to the attended
transfer ARI test in the test suite.
(closes issue ASTERISK-23287)
Reported by: Matt Jordan
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Added the ability for transferring directly to voicemail on digium phones.
Added a new module that checks for the presence of a custom header and/or
diversion header within a sip REFER. If either is found and they specify
a sending to voicemail action then variables are added to the channel
allowing the user access to them in the dialplan. Dialplan can then be
written that branches based upon these values allowing, for instace, for
a single number to be used for dialing and/or accessing voicemail directly.
Also fixed a problem where the PJSIP_HEADER function was allowing non pjsip
channels through (checked to make sure it has the correct channel type before
proceeding).
Review: https://reviewboard.asterisk.org/r/3245/
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* Fixed off-nominal json ref counting issue with using the following API
calls: ast_json_object_set() and ast_json_array_append().
* Fixed off-nominal error reporting in ast_ari_endpoints_list().
* Fixed some miscellaneous off-nominal json ref counting issues in
report_receive_fax_status() and dial_to_json().
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In order to retrieve an arbitrary sorcery instance from a dialplan function
(or any place else) there needs to be a registry of sorcery instances.
ast_sorcery_init now creates a hashtab as a registry.
ast_sorcery_open now checks the hashtab for an existing sorcery instance
matching the caller's module name. If it finds one, it bumps the
refcount and returns it. If not, it creates a new sorcery instance,
adds it to the hashtab, then returns it.
ast_sorcery_retrieve_by_module_name is a new function that does a hashtab
lookup by module name. It can be called by the future dialplan function.
res_pjsip/config_system needed a small change to share the main res_pjsip
sorcery instance.
tests/test_sorcery was updated to include a test for the registry.
(closes issue ASTERISK-22537)
Review: http://reviewboard.asterisk.org/r/3184/
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When 'use_avpf' is set to True, inbound offers must use the AVPF/SAVPF RTP
profile. However, when 'use_avpf' is set to False, Asterisk will accept
both AVP/SAVP or AVPF/SAVPF RTP profiles in inbound offers. The documentation
previously implied that Asterisk would reject AVPF/SAVPF if 'use_avpf' was
set to False and a UA offered said profile in an INVITE request.
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The sorcery astDB wizzard does not handle regex correctly if the pattern
begins with an anchor character.
This patch attempts to convert the anchored regex pattern to a prefix
pattern supported by astDB for performance reasons. If it is not able to
convert the pattern it falls back to getting all astDB members of the
family and doing a normal regex pattern matching on the retrieved records.
Review: https://reviewboard.asterisk.org/r/3161/
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If an AOR has no permanent contacts, then the
permanent_contacts container is never allocated.
This makes the code safe in the face of NULLs.
I also changed the variable that counts contacts
from "num" to "total_contacts" since there are now
two variables that are indicate numbers of things.
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This patch tweaks the behaviour of POST /channels with channel variables such
that the variables are passed into the pbx.c routines that perform the
origination. This allows the variables to be assigned to the newly created
channels immediately upon their construction, as opposed to be assigned after
the originate has completed.
The upshot of this is that the variables are available on the channels if
they execute in the dialplan, as opposed to only being available once the
channels are answered.
Review: https://reviewboard.asterisk.org/r/3183/
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This change allows timing implementation data to be stored directly
on the timer itself thus removing the requirement for many
implementations to do a container lookup for the same information.
This means that API calls into timing implementations can directly
access the information they need instead of having to find it.
Review: https://reviewboard.asterisk.org/r/3175/
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* Adds identify, transport, and registration support to the PJSIP CLI.
* Creates three additional callbacks, one for an iterator, one for a
comparator, and one for a container. This eliminates the link dependency
from higher level modules to lower level ones.
* Eliminates duplicate sorting in PJSIP CLI commands.
* Cleans up PJSIP CLI output formatting.
* Pushes CLI command registration down to the implementing source file.
* Adds several ast_sip_destroy_sorcery functions to complement existing
ast_sip_sorcery_initialize functions. The destroy functions unregister
PJSIP CLI commands and PJSIP CLI formatters.
Reported by: George Joseph
Review: https://reviewboard.asterisk.org/r/3104/
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If the global section was not specified in pjsip.conf then the configuration
object does not exist in sorcery so when retrieving "debug" option it would
return NULL. Then the NULL result was passed to ast_false utils function
which would return false because it wasn't set to some representation of
false, thus enabling sip debug logging. Made it so if the global config object
does not exist then it will return a default of "no" for sip debugging.
(issue ASTERISK-23038)
Reported by: Rusty Newton
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The code assumed that unregistering the alias would always succeed while in
practice this is not actually true. A common case is the "reload" command itself.
If the cli_aliases.conf configuration file was changed and reload executed the
command would fail to unregister and ultimately point to freed memory.
The reload process now checks whether unregistering succeeded or not and if not
the old CLI alias is retained.
(closes issue ASTERISK-19773)
Reported by: Joel Vandal
(closes issue ASTERISK-22757)
Reported by: Gareth Blades
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When the PJSIP pubsub framework was created, subscription handlers were required
to state what event they handled along with what body types they knew how to
generate. While this serves well when implementing a base RFC, it has problems
when trying to extend the body to support non-standard or proprietary body
elements. The code also was NOTIFY-specific, meaning that when the time comes
that we start writing code to send out PUBLISH requests with MWI or presence
bodies, we would likely find ourselves duplicating code that had previously been
written.
This changeset introduces the concept of body generators and body supplements. A
body generator is responsible for allocating a native structure for a given body
type, providing the primary body content, converting the native structure to a
string, and deallocating resources. A body supplement takes the primary body
content (the native structure, not a string) generated by the body generator and
adds nonstandard elements to the body. With these elements living in their own
module, it becomes easy to extend our support for body types and to re-use
resources when sending a PUBLISH request.
Body generators and body supplements register themselves with the pubsub core,
similar to how subscription and publish handlers had done. Now, subscription
handlers do not need to know what type of body content they generate, but they
still need to inform the pubsub core about what the default body type for a
given event package is. The pubsub core keeps track of what body generators and
body supplements have been registered. When a SUBSCRIBE arrives, the pubsub core
will check that there is a subscription handler for the event in the SUBSCRIBE,
then it will check that there is a body generator that can provide the content
specified in the Accept header(s).
Because of the nature of body generators and supplements, it means
res_pjsip_exten_state and res_pjsip_mwi have been completely gutted. They no
longer worry about body types, instead calling
ast_sip_pubsub_generate_body_content() when they need to generate a NOTIFY body.
Review: https://reviewboard.asterisk.org/r/3150
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When subscribing to MWI (res_pjsip_mwi) and the sip uri did not contain a name
(ex: sip:<ip address>) then the subscription would fail since it would be unable
to locate an associated aor. This patch makes it so that when a subscribe comes
with no aor name then it will subscribe to all aors on the located endpoint.
(closes issue ASTERISK-23072)
Reported by: Bob M
Review: https://reviewboard.asterisk.org/r/3164/
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In NAT scenarios where a call is placed to a Grandstream phone,
res_pjsip will sometimes send the ACK to a 200 OK to the private
address of the device behind the NAT instead of the address of the NAT
device. This corrects that behavior by rewriting the address in the
Contact header in the incoming 200 OK and the dialog's target address
if necessary (since it has already been rewritten to the incorrect
private address).
(closes issue ASTERISK-23106)
Review: https://reviewboard.asterisk.org/r/3168/
Reported by: Matt Jordan
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What seems to be happening is if a subscription has been terminated and the
subscription timeout/expires is less than the time it takes for all pending
transactions (currently on the subscription) to end then the subscription
timer will not have been canceled yet and sub will be null. Since the
subscription has already been canceled nothing needs to be done so a null
check in the asterisk code is sufficient in working around this problem.
(closes issue ASTERISK-23129)
Reported by: Dan Jenkins
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On some systems the values for INFINITY and NAN are not defined thus causing
a build error on those systems. Added definitions for those if they had
not previously been defined.
(closes issue ASTERISK-23056)
Reported by: capouch
Patches:
inf-nan-patch.txt uploaded by capouch (license 6564)
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Currently, attempting to subscribe an application to a device state
that it has already subscribed to will generate a 500 error response.
This will now be treated as a subscription refresh even though ARI
subscriptions don't currently support lifetimes and will respond with
the normal response for a successful subscription (200 OK).
(closes issue ASTERISK-23143)
Reported by: Matt Jordan
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The core may (depending on circumstances) request a single codec on outgoing
calls. Many channel drivers ignore or treat this as a suggestion while still
including configured codecs. The res_pjsip_session logic treated this as
an explicit request, leaving out other configured codecs.
This change makes res_pjsip_session behave like other channel driver and simply
adds the requested codec to the list.
(closes issue ASTERISK-23082)
Reported by: xrobau
Review: https://reviewboard.asterisk.org/r/3140/
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r406294 | mmichelson | 2014-01-23 15:00:24 -0600 (Thu, 23 Jan 2014) | 11 lines
Fix presence body errors found during testing:
* PIDF bodies were reporting an "open" state in many cases where
it should have been reporting "closed"
* XPIDF bodies had XML nodes placed incorrectly within the hierarchy.
* SIP URIs in XPIDF bodies did not go through XML sanitization
* XML sanitization had some errors:
* Right angle bracket was being replaced with "&rt;" instead of ">"
* Double quote, apostrophe, and ampersand were not being escaped.
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r406295 | mmichelson | 2014-01-23 15:09:35 -0600 (Thu, 23 Jan 2014) | 11 lines
Fix presence body errors found during testing:
* PIDF bodies were reporting an "open" state in many cases where
it should have been reporting "closed"
* XPIDF bodies had XML nodes placed incorrectly within the hierarchy.
* SIP URIs in XPIDF bodies did not go through XML sanitization
* XML sanitization had some errors:
* Right angle bracket was being replaced with "&rt;" instead of ">"
* Double quote, apostrophe, and ampersand were not being escaped.
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Several of the playback error messages for invalid media input in
res_stasis_playback.c had the media name and channel name reversed.
They now correctly identify the channel name and media name.
Reported by: skrusty
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Making the help text for both more explicit regarding the format of mailbox identifiers. i.e. clarifying the format for app_voicemail mailboxes vs mailboxes from external MWI sources through modules such as res_external_mwi.
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The PJSIP header parsing function (pjsip_parse_hdr) can generate more
than one header instance from a single header field. These header
instances exist as a list attached to the returned header and must be
handled appropriately when they are added to a message or else only the
first header instance will be used. This changes the linked list
functions used in outbound proxy code to merge the lists properly.
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This adds back in support for specifying channel variables during an
originate without compromising the ability to specify query parameters
in the JSON body. This was accomplished by generating the body-parsing
code in a separate function instead of being integrated with the URI
query parameter parsing code such that it could be called by paths with
body parameters. This is transparent to the user of the API and
prevents manual duplication of code or data structures.
(closes issue ASTERISK-23051)
Review: https://reviewboard.asterisk.org/r/3122/
Reported by: Matt Jordan
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This change adds improvements to support for allow=all in
pjsip.conf so that it functions as intended. Previously,
the allow/disallow socery configuration would set & clear
codecs from the media.codecs and media.prefs list, but if
all was specified the prefs list was not updated. Then a
call would fail when create_outgoing_sdp_stream() created
an SDP with no audio codecs.
A new function ast_codec_pref_append_all() is provided to
add all codecs to the prefs list - only those not already
on the list. This enables the configuration to specify a
codec preference, but still add all codecs, and even then
remove some codecs, as shown in this example:
allow = ulaw, alaw, all, !g729, !g723
Also, the display order of allow in cli output is updated
to match the configuration by using prefs instead of caps
when generating a human readable string.
Finally, a change to create_outgoing_sdp_stream() skips a
codec when it does not have a payload code instead of the
call failing.
(closes issue ASTERISK-23018)
Reported by: xrobau
Review: https://reviewboard.asterisk.org/r/3131/
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If an endpoint had previously dynamically registered a contact and the contact
information was successfully stored in astdb then upon restart the qualify
notifications would not be sent out if the qualify_frequency was set. This was
due to the fact that only permanent contacts were being checked and scheduled
for qualifies on startup. Modified the code to check and schedule all
registered contacts at startup.
(closes issue ASTERISK-23062)
Reported by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/3124/
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When path support was added and contacts were made available during
request creation and transmission, the code path used by outbound
qualify support was not modified correctly and was causing request
creation to fail. This ensures that outbound request creation with only
a contact and no dialog, endpoint, or uri can succeed which restores
qualify support.
Reported by: gtjoseph
Reported by: kharwell
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This adds Path support to chan_pjsip in res_pjsip_path.c with minimal
additions in res_pjsip_registrar.c to store the path and additions in
res_pjsip_outbound_registration.c to enable advertisement of path
support to registrars and intervening proxies.
Path information is stored on contacts and is enabled via Address of
Record (AoRs) and Registration configuration sections.
While adding path support, it became necessary to be able to add SIP
supplements that handled messages outside of sessions, so a framework
for handling these types of hooks was added in parallel to the
already-existing session supplements and several senders of
out-of-dialog requests were refactored as a result.
(closes issue ASTERISK-21084)
Review: https://reviewboard.asterisk.org/r/3050/
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Adds the following AMI commands:
PUT mailboxes/mailboxName
modifies mailbox state and implicitly creates new mailboxes
GET mailboxes/mailboxName
retrieves a JSON representation of a single mailbox if it exists
GET mailboxes
retrieves a JSON array of all mailboxes
DELETE mailbox/mailboxName
deletes a mailbox
Note that res_mwi_external must be loaded for these functions to
actually do anything.
Review: https://reviewboard.asterisk.org/r/3117/
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We were not including an authentication challenge when sending a 401 response
to unmatched endpoints. This was due to the conversion to use a vector for
authentication section names on an endpoint. The vector for artificial endpoints
was empty, resulting in the challenge being sent back containing no challenges.
This is worked around by placing a bogus value in the artificial endpoint's auth
vector. This value is never looked up by anything, since they instead will directly
call ast_sip_get_artificial_auth().
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Matthew Jordan
Joshua Colp
Kinsey Moore
Kevin Harwell
Corey Farrell
Richard Mudgett
George Joseph
Mark Michelson
Walter Doekes
Jonathan Rose
Rusty Newton
Scott Griepentrog