Asterisk didn't support the dynamic payload change in rtp mapping in the 200
OK response.
Scenario:
Asterisk sends the INVITE proposing alaw and telephone-event, it proposes
rtpmap:101 for telephone-event. Peer responds with 2xx, it answers with
alaw and telephone-event also, but it proposes a different rtpmap number
(rtpmap:103) for telephone-event.
Expected Behaviour:
Asterisk should honour the rtpmapping in the response and send DTMF packets
using 103 as payload type for DTMF.
Actual Behaviour: Asterisk sends DTMF packets using payload type 101.
With this patch asterisk now supports changes that can occur in the rtp mapping
in the response.
(closes issue ASTERISK-23279)
Reported by: NITESH BANSAL
Review: https://reviewboard.asterisk.org/r/3225/
Patches:
dynamic_payload_change.patch uploaded by nbansal (license 6418)
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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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Updated the code to check to see if MOH is playing on the transferor and if
so then start it on the channel that replaces it during a masquerade.
Example scenario of the problem:
Alice calls Bob and then Bob begins the attended transfer process into a queue.
Upon going on hold Alice hears music and so does Bob once he is in the queue.
Bob then transfers Alice into the queue and then music for Alice stops even
though she should be hearing it since has now replaced Bob in the queue.
The problem that was occurring is that once the channel was masqueraded the app
(queues, confbridge, etc...) had no way of knowing that the channel had just
been swapped out thus it did not start music for the present channel.
Credit to Olle Johansson for pointing me in the right direction on this issue.
(closes issue ASTERISK-19499)
Reported by: Timo Teräs
Review: https://reviewboard.asterisk.org/r/3226/
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When using the "x" option (specify a DTMF digit to exit the application), it is
not obvious in the documentation that this only works when spying on a channel.
If a channel being used to spy on other channels is waiting to connect to a
channel or is no longer attached to a channel, the DTMF is ignored.
As noted on the issue tracker, since there are workarounds available and this is
a rarely used option we are opting for a documentation change here.
(closes issue ASTERISK-22661)
Reported by: Chris Hillman
Patches:
asterisk-22661-doc-clarify-chan_spy.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2990/
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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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It is highly unlikely, but - at least in Asterisk 12 - theoretically possible
to load Asterisk with no dialplan whatsoever. If that occurs, and some other
module (that is not a pbx module) attempts to merge its contexts into the
dialplan, the existing merge routine will crash. This is because it is not
insane, and rightly believes that you provided some sort of dialplan,
somewhere.
This patch will gracefully merge the contexts in such a case. Note that this
is highly unlikely to occur in 1.8/11, as features will most likely provide
some dialplan via parking. However, in Asterisk 12, parking is now provided
by res_parking, and hence may create its dialplan later.
(closes issue ASTERISK-23297)
Reported by: CJ Oster
Review: https://reviewboard.asterisk.org/r/3222
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URI's are supposed to be case sensitive and all
lower case. In practice some portions of URI's
in ARI are case insensitive and others are not,
such as TECH, which in one instance would match
a lower case name and in another would not. In
this patch, the ast_endpoint_lastest_snapshot()
function is modified to change the TECH portion
to full upper case before lookup. This resolves
the discrepancy noted by the reporter. However
I chose to avoid forcing the /ari prefix of the
URI's to be lower case for now. Except for the
two cases here, all URI's should be lower case,
unless they are part of a resource name or id.
Review: https://reviewboard.asterisk.org/r/3211/
Reported by: Zane Conkle
(closes issue ASTERISK-23125)
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In ast_format_sdp_parse and ast_format_sdp_generate
the check checks for a valid interface and function
were potentially confusing, and hid an error in the
test of the presence of the function that is called
later. This patch clears up and corrects the test.
Review: https://reviewboard.asterisk.org/r/3208/
(closes issue ASTERISK-23098)
Reported by: marcelloceschia
Patches:
main_format.patch uploaded by marcelloceschia (license 6036)
ASTERISK-23098.patch uploaded by coreyfarrell (license 5909)
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This patch moves setting SIP_DEFER_BY_ON_TRANSFER prior to calling
ast_bridge_transfer_blind. This prevents a BYE from being sent prior to the
NOTIFY request that informs the transferor if the transfer succeeded or failed.
This patch also clears said flag from the off nominal NOTIFY paths in the
local_attended_transfer code, as once we've sent the NOTIFY request it is safe
to send by the BYE request.
This was caught by the blind-transfer-accountcode test in the Asterisk Test
Suite.
(closes issue ASTERISK-23290)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3214/
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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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The old code depended on undefined va_arg behaviour: calling a function
twice with the same va_list parameter and expecting it to continue where
it left off. The changed code behaves like the manpage says it should.
Also added a bunch of early returns to trap errors (e.g. OOM) instead of
crashing.
The problem was found by Julian Lyndon-Smith. The deviant behaviour on
the raspberry PI also uncovered another bug (fixed in r407875) in the
res_config_pgsql.so driver.
Reported by: jmls
Tested by: jmls
Review: https://reviewboard.asterisk.org/r/3201/
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This is a first stab at tweaking the performance profile of the scheduler. Removing
the hashtab usage removes an extra memory allocation when scheduling something and
makes it so rescheduling does not incur any memory allocation at all.
Review: https://reviewboard.asterisk.org/r/3199/
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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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* Move route code to sip/route.c + sip/include/route.h
* Rename functions to sip_route_*
* Replace ad-hoc list code with macro's from linkedlists.h
* Create sip_route_process_header() to processes Path and Record-Route headers
(previously done with different code in build_route and build_path)
* Add use of const where possible
* Move struct uriparams, struct contact and contactliststruct from sip.h to
reqresp_parser.h. sip/route.c uses reqresp_parser.h but not sip.h, this was
a problem. These moved declares are not used outside of reqresp_parser.
* While modifying reqprep() the lack of {} caused me trouble. I added them.
* Code outside route.c treats sip_route as an opaque structure, using macro's
or procedures for all access.
(closes issue ASTERISK-22582)
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/3173/
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Currently, when the first marked user enters the conference that
contains waitmarked users, a prompt is played indicating that the user
is being placed into the conference. Unfortunately, this prompt is
played to the marked user and not the waitmarked users which is not
very helpful.
This patch changes that behavior to play a prompt stating
"The conference will now begin" to the entire conference after adding
and unmuting the waitmarked users since the design of confbridge is not
conducive to playing a prompt to a subset of users in a conference in
an asynchronous manner.
(closes issue PQ-1396)
Review: https://reviewboard.asterisk.org/r/3155/
Reported by: Steve Pitts
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When formatting an optional IE, the value is, of course, optional. As such, it
is entirely appropriate for ast_json_object_get to return NULL. If that occurs,
we now simply skip the IE that was requested, as it was not provided by the
entity that raised the event.
Thanks to George Joseph (gtjoseph) for catching this and reporting it in
#asterisk-dev
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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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When extracting timestamps that are parsed, time stamp values that are not set
(time values of 0.000000) should not actually result in a parsed string. The
value should be skipped, and the result of the CDR function should be an
empty string.
Prior to this patch, the result was fed to the time formatting, which would
result in an output of a date/time in 1969.
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Establishing an IAX2 call between Asterisk v1.4 and v1.8 (or later)
results in an unexpected call disconnect. The problem happens because
newer values in the enum ast_control_frame_type are not consistent between
the branch versions of Asterisk.
For example:
1) v1.4 calls v1.8 (or later) using IAX2
2) v1.8 answers and sends a connected line update control frame. (on v1.8
AST_CONTROL_CONNECTED_LINE = 22)
3) v1.4 receives the control frame as an end-of-q (on v1.4
AST_CONTROL_END_OF_Q = 22)
4) v1.4 disconnects the call once the receive queue becomes empty.
Several things are done by this patch to fix the problem and attempt to
prevent it from happening again in the future:
* Added a warning at the definition of enum ast_control_frame_type about
how to add new control frame values.
* Made block sending and receiving control frames that have no reason to
go over the wire.
* Extended the connectedline iax.conf parameter to also include the
redirecting information updates.
* Updated the connectedline iax.conf parameter documentation to include a
notice that the parameter must be "no" when the peer is an Asterisk v1.4
instance.
(closes issue AST-1302)
Review: https://reviewboard.asterisk.org/r/3174/
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The appdocsxml.dtd specifies that a "required" attribute in a parameter may
have a value of yes, no, true, or false. On some systems, specifying "False"
instead of "false" would cause a validation error. This patch fixes the casing
to explicitly match the DTD.
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If an enum had been previously created the alembic script would attempt to
re-create it and an error would be generated while running migrations for a
postgresql server. The work around for this is to use the ENUM object type
for postgres as opposed to the generic enum type used by sqlalchemy. Using
this type in the script seems to work properly for both postgres and mysql.
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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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The changes log was written with language that was a little too internal
Asterisk specific, so it's been changed to be more in the frame of reference
of an ARI user. Also, previously the AMI event changes were omitted from the
change log as well as the ability to include a bridge name in the ARI post
bridges command.
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Corey Farrell
Kevin Harwell
Richard Mudgett
Alexandr Anikin
Michael L. Young
George Joseph
Matthew Jordan
Rusty Newton
Mark Michelson
Scott Griepentrog
Walter Doekes
Tzafrir Cohen
Kinsey Moore
Joshua Colp
Jonathan Rose