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Commit 1bce690ccb36a4744a327c07af23a9a3a0fa20cd was incrementing txcount
for rtcp packets as well as rtp packets and that was causing sender reports
to be generated instead of receiver reports in cases where no rtp was actually
being sent.
Moved the txcount increment from __rtp_sento, which handles both rtp and rtcp,
to rtp_sento which only handles rtp packets.
Discovered by the hep/rtcp-receiver test.
Change-Id: Ie442e4bb947a68847a676497021ba10ffaf376d5
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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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No one seemed to notice but every time an OPTIONS goes out, it goes
out with a From of "asterisk" (or whatever the default from_user is set to),
even if you specify an endpoint.
The issue had several causes...
qualify_contact is only called with an endpoint if called from the CLI.
If the endpoint is NULL, qualify_contact only looks up the endpoint if
authenticate_qualify=yes. Even then, it never passes it on to
ast_sip_create_request where the From header is set. Therefore From
is always "asterisk" (or whatever the default from_user is set to).
Even if ast_sip_create_request were to get an endpoint, it only sets
the From if endpoint->from_user is set.
The fix is 4 parts...
First, create_out_of_dialog_request was modified to use the endpoint id
if endpoint was specified and from_user is not set.
Second, qualify_contact was modified to always look up an endpoint if
one wasn't specified regardless of authenticate_qualify. It then passes
the endpoint on to create_out_of_dialog_request.
Third (and most importantly), find_an_endpoint was modified to find
an endpoint by using an "aors LIKE %contact->aor%" predicate with
ast_sorcery_retrieve_by_fields. As such, this patch will only work
if the sorcery realtime optimizations patch goes in. Otherwise we'd
be pulling the entire endpoints database every time we send an OPTIONS.
Since we already know the contact's aor, the on_endpoint callback was also
modified to just check if the contact->aor is an exact match to one of
the endpoint's.
Finally, since we now have an endpoint for every OPTIONS request,
res_pjsip/endpt_send_request (which handles out-of-dialog reqests) was
updated to get the transport from the endpoint and set it on tdata.
Now the correct transport is used.
Change-Id: I2207e12bb435e373bd1e03ad091d82e5aba011af
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Asterisk uses separate UDP ports for RTP and RTCP traffic and RFC 5764
explicitly states:
There MUST be a separate DTLS-SRTP session for each distinct pair of
source and destination ports used by a media session
This means RTP keying material cannot be used for DTLS RTCP, which was
the reason why RTCP encryption would fail.
ASTERISK-25642
Change-Id: I7e8779d8b63e371088081bb113131361b2847e3a
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rxcount, txcount, rxoctetcount and txoctetcount weren't being calculated
for bridged streams because the calulations were being done after the
bridged short-circuit. Actually, rxoctetcount wasn't ever being calculated.
Moved the calculations so they occur for all valid received packets and
all transmitted packets. Also added rxoctetcount and txoctetcount to
ast_rtp_instance_stat.
Change-Id: I08fb06011a82d38c3b4068867a615068fbe59cbb
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Blind transfers to a recognized parking extension need to use the parker's
channel variable values to create the dynamic parking lot. This is
because there is always only one parker while the parkee may actually be a
multi-party bridge. A multi-party bridge can never supply the needed
channel variables to create the dynamic parking lot. In the multi-party
bridge blind transfer scenario, the parker's CHANNEL(parkinglot) value and
channel variables are inherited by the local channel used to park the
bridge.
* In park_common_setup(), make use the parker instead of the parkee to
supply the dynamic parking lot channel variable values. In all but one
case, the parkee is the same as the parker. However, in the recognized
parking extension blind transfer scenario for a two party bridge they are
different channels. For consistency, we need to use the parker channel.
* In park_local_transfer(), pass the CHANNEL(parkinglot) value to the
local channel when blind transferring a multi-party bridge to a recognized
parking extension.
* When a local channel starts a call, the Local;2 side needs to inherit
the CHANNEL(parkinglot) value from Local;1.
The DTMF one-touch parking case wasn't even trying to create dynamic
parking lots before it aborted the attempt.
* In parking_park_call(), add missing code to create a dynamic parking
lot.
A DTMF bridge hook is documented as returning -1 to remove the hook.
Though the hook caller is really coded to accept non-zero. See the
ast_bridge_hook_callback typedef.
* In feature_park_call(), don't remove the DTMF one-touch parking hook
because of an error.
ASTERISK-24605 #close
Reported by: Philip Correia
Patches:
call_park.patch (license #6672) patch uploaded by Philip Correia
Change-Id: I221d3a8fcc181877a1158d17004474d35d8016c9
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There were a number of places in the res_pjsip stack that were getting
all endpoints or all aors, and then filtering them locally.
A good example is pjsip_options which, on startup, retrieves all
endpoints, then the aors for those endpoints, then tests the aors to see
if the qualify_frequency is > 0. One issue was that it never did
anything with the endpoints other than retrieve the aors so we probably
could have skipped a step and just retrieved all aors. But nevermind.
This worked reasonably well with local config files but with a realtime
backend and thousands of objects, this was a nightmare. The issue
really boiled down to the fact that while realtime supports predicates
that are passed to the database engine, the non-realtime sorcery
backends didn't.
They do now.
The realtime engines have a scheme for doing simple comparisons. They
take in an ast_variable (or list) for matching, and the name of each
variable can contain an operator. For instance, a name of
"qualify_frequency >" and a value of "0" would create a SQL predicate
that looks like "where qualify_frequency > '0'". If there's no operator
after the name, the engines add an '=' so a simple name of
"qualify_frequency" and a value of "10" would return exact matches.
The non-realtime backends decide whether to include an object in a
result set by calling ast_sorcery_changeset_create on every object in
the internal container. However, ast_sorcery_changeset_create only does
exact string matches though so a name of "qualify_frequency >" and a
value of "0" returns nothing because the literal "qualify_frequency >"
doesn't match any name in the objset set.
So, the real task was to create a generic string matcher that can take a
left value, operator and a right value and perform the match. To that
end, strings.c has a new ast_strings_match(left, operator, right)
function. Left and right are the strings to operate on and the operator
can be a string containing any of the following: = (or NULL or ""), !=,
>, >=, <, <=, like or regex. If the operator is like or regex, the
right string should be a %-pattern or a regex expression. If both left
and right can be converted to float, then a numeric comparison is
performed, otherwise a string comparison is performed.
To use this new function on ast_variables, 2 new functions were added to
config.c. One that compares 2 ast_variables, and one that compares 2
ast_variable lists. The former is useful when you want to compare 2
ast_variables that happen to be in a list but don't want to traverse the
list. The latter will traverse the right list and return true if all
the variables in it match the left list.
Now, the backends' fields_cmp functions call ast_variable_lists_match
instead of ast_sorcery_changeset_create and they can now process the
same syntax as the realtime engines. The realtime backend just passes
the variable list unaltered to the engine. The only gotcha is that
there's no common realtime engine support for regex so that's been noted
in the api docs for ast_sorcery_retrieve_by_fields.
Only one more change to sorcery was done... A new config flag
"allow_unqualified_fetch" was added to reg_sorcery_realtime.
"no": ignore fetches if no predicate fields were supplied.
"error": same as no but emit an error. (good for testing)
"yes": allow (the default);
"warn": allow but emit a warning. (good for testing)
Now on to res_pjsip...
pjsip_options was modified to retrieve aors with qualify_frequency > 0
rather than all endpoints then all aors. Not only was this a big
improvement in realtime retrieval but even for config files there's an
improvement because we're not going through endpoints anymore.
res_pjsip_mwi was modified to retieve only endpoints with something in
the mailboxes field instead of all endpoints then testing mailboxes.
res_pjsip_registrar_expire was completely refactored. It was retrieving
all contacts then setting up scheduler entries to check for expiration.
Now, it's a single thread (like keepalive) that periodically retrieves
only contacts whose expiration time is < now and deletes them. A new
contact_expiration_check_interval was added to global with a default of
30 seconds.
Ross Beer reports that with this patch, his Asterisk startup time dropped
from around an hour to under 30 seconds.
There are still objects that can't be filtered at the database like
identifies, transports, and registrations. These are not going to be
anywhere near as numerous as endpoints, aors, auths, contacts however.
Back to allow_unqualified_fetch. If this is set to yes and you have a
very large number of objects in the database, the pjsip CLI commands
will attempt to retrive ALL of them if not qualified with a LIKE.
Worse, if you type "pjsip show endpoint <tab>" guess what's going to
happen? :) Having a cache helps but all the objects will have to be
retrieved at least once to fill the cache. Setting
allow_unqualified_fetch=no prevents the mass retrieve and should be used
on endpoints, auths, aors, and contacts. It should NOT be used for
identifies, registrations and transports since these MUST be
retrieved in bulk.
Example sorcery.conf:
[res_pjsip]
endpoint=config,pjsip.conf,criteria=type=endpoint
endpoint=realtime,ps_endpoints,allow_unqualified_fetch=error
ASTERISK-25826 #close
Reported-by: Ross Beer
Tested-by: Ross Beer
Change-Id: Id2691e447db90892890036e663aaf907b2dc1c67
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Change-Id: I8f7a8890aef27824301c642d4d15407ac83e6f02
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res/parking/parking_applications.c:
* Add malloc fail checks in setup_park_common_datastore().
* Fix playing parking failed announcement to only happen on non-blind
transfers in park_app_exec(). It could never go out before because a test
was provedly always false.
res/parking/parking_bridge.c:
* Fix NULL tolerance in generate_parked_user() because
bridge_parking_push() can theoretically pass a NULL parker channel if the
parker channel went away for some reason.
* Clarify some weird code dealing with blind_transfer in
bridge_parking_push().
res/parking/parking_bridge_features.c:
* Made park_local_transfer() set BLINDTRANSFER on the Local;1 channel
which will be bulk copied to the Local;2 channel on the subsequent
ast_call(). The additional advantage is if the parker channel has the
BLINDTRANSFER and ATTENDEDTRANSFER variables set they are now guaranteed
to be overridden.
res/parking/parking_manager.c:
* Fix AMI Park action input range checking of the Timeout header in
manager_park().
* Reduced locking scope to where needed in manager_park().
res/res_parking.c:
* Fix some off nominal missing unlocks by eliminating the returns.
Change-Id: Ib64945bc285acb05a306dc12e6f16854898915ca
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* Remove duplicate res_parking.conf courtesytone config option
documentation.
ASTERISK-24596 #close
Reported by: Philip Correia
ASTERISK-24605
Reported by: Philip Correia
Patches:
call_park_app_doc.patch (license #6672) patch uploaded by Philip Correia
Change-Id: I90a92a891c6494dc08173e675856afcc4764c5b5
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database." into 13
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The configuration unsigned integer option handler sets flags for the
parser as if the option should be a signed integer (PARSE_INT32),
leading to errors on "out of range" values. Fix flags (PARSE_UINT32).
A fix to res_pjsip is also present which stops invalid flags from
being passed when registering sorcery object fields for qualify
status.
ASTERISK-25612 #close
Change-Id: I96b539336275e0e72a8e8033487d2c3344debd3e
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The log message when a MusicOnHold music class was not found was changed
from debug level to WARNING level in Asterisk 11.19 and 13.5. For those
using realtime musiconhold, this message is wrong because it warns
before checking the database.
This changeset delays the warning until after the database has been
checked.
Reported-by: Conrad de Wet
ASTERISK-25444 #close
Change-Id: I6cfb2db2f9cfbd2bb3d30566ecae361c4abf6dbf
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The "Refer-to" header of an incoming REFER request is parsed by
pjsip_parse_uri(). That function requires the URI parameter to be NULL
terminated. Unfortunately, the previous code added the NULL terminator by
overwriting memory that may not be safe. The overwritten memory results
could be benign, memory corruption, or a segmentation fault. Now the URI
is NULL terminated safely by copying the URI to a new chunk of memory with
the correct size to be NULL terminated.
ASTERISK-25814 #close
Change-Id: I32565496684a5a49c3278fce06474b8c94b37342
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setting." into 13
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100" into 13
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Older versions of PJSIP do not have the proto field on the TLS transport
setting structure. This change adds a configure check so even if it is
not present we will still be able to build.
Change-Id: Ibf3f47befb91ed1b8194bf63888baa6fee05aba9
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prohibited" into 13
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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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ast_sip_get_transport_states" into 13
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Per RFC3325, the 'From' header is now anonymized on outgoing calls when
caller id presentation is prohibited.
TID = trust_id_outbound
PRO = Set(CALLERID(pres)=prohib)
USR = endpoint/from_user
DOM = endpoint/from_domain
PAI = YES(privacy=off), NO(not sent), PRI(privacy=full) (assumes send_pai=yes)
Conditions |Result
--------------------|----------------------------------------------------
TID PRO USR DOM |PAI FROM
--------------------|----------------------------------------------------
Y Y abc def.ghi |PRI "Anonymous" <sip:abc@def.ghi>
Y Y abc |PRI "Anonymous" <sip:abc@anonymous.invalid>
Y Y def.ghi |PRI "Anonymous" <sip:anonymous@def.ghi>
Y Y |PRI "Anonymous" <sip:anonymous@anonymous.invalid>
Y N abc def.ghi |YES <sip:abc@def.ghi>
Y N abc |YES <sip:abc@<ip_address>>
Y N def.ghi |YES "Caller Name" <sip:<caller_exten>@def.ghi>
Y N |YES "Caller Name" <sip:<caller_exten>@<ip_address>>
N Y abc def.ghi |NO "Anonymous" <sip:abc@def.ghi>
N Y abc |NO "Anonymous" <sip:abc@anonymous.invalid>
N Y def.ghi |NO "Anonymous" <sip:anonymous@def.ghi>
N Y |NO "Anonymous" <sip:anonymous@anonymous.invalid>
N N abc def.ghi |YES <sip:abc@def.ghi>
N N abc |YES <sip:abc@<ip_address>>
N N def.ghi |YES "Caller Name" <sip:<caller_exten>@def.ghi>
N N |YES "Caller Name" <sip:<caller_exten>@<ip_address>>
ASTERISK-25791 #close
Reported-by: Anthony Messina
Change-Id: I2c82a5ca1413c2c00fb62ea95b0ae8e97af54dc9
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During the transfer process, some phones (okay it was the Jitsi softphone,
but maybe others are out there) send a "bye" immediately after receiving a
SIP Notify. When a "bye" is received early for some types of transfers the
transferer channel may no longer be available during late stage transfer
processing.
For instance, during an attended transfer involving stasis bridging at one
point the created local channel looks for an associated swap channel in
order to retrieve the stasis application name. If the transferer has hung
up then the local channel will fail to find it. The local channel then has
no way to know which stasis app to enter, so it fails and hangs up as well.
Thus the transfer does not complete as expected.
This patch delays the sending of the initial notify in order to give the
transfer process enough time to gather the necessary data for a successful
transfer.
ASTERISK-25771
Change-Id: I09cfc9a5d6ed4c007bc70625e0972b470393bf16
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PJSIP does not ensure that when printing the message body the
buffer will be NULL terminated. This is problematic when searching
for the signal and duration values of the DTMF.
This change ensures the buffer is always NULL terminated.
Change-Id: I52653a1a60c93092d06af31a27408d569cc98968
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into 13
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reason." into 13
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ast_sip_get_transport_states was returning a container of internal_state
objects instead of ast_sip_transport_state objects. This was causing
transport lookups to fail, most noticably in res_pjsip_nat, which
couldn't find the correct external addresses. This was causing contacts
to go out with internal ip addresses.
ASTERISK-25830 #close
Reported-by: Sean Bright
Change-Id: I1aee6a2fd46c42e8dd0af72498d17de459ac750e
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In message.c, if msg_alloc fails to init the string field,
vars may be null, so use a null tolerant cleanup.
In res_pjsip_messaging.c, if msg_data_create fails, mdata
will be null, so use a null tolerant cleanup.
ASTERISK-25323
Change-Id: Ic2d55c2c3750d5616e2a05ea92a19c717507ff56
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This patch avoids crashing on a null pointer
if the strdup() allocation fails.
ASTERISK-25323
Change-Id: I3f67434820ba53b53663efd6cbb42749f4f6c0f5
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Previous chan_sip behavior:
Before this patch chan_sip would always strip any quotes from an incoming
reason and pass that value up as the REDIRECTING(reason). For an outgoing
reason value, chan_sip would check the value against known values and
quote any it didn't recognize. Incoming 480 response message reason text
was just assigned to the REDIRECTING(reason).
Previous chan_pjsip behavior:
Before this patch chan_pjsip would always pass the incoming reason value
up as the REDIRECTING(reason). For an outgoing reason value, chan_pjsip
would send the reason value as passed down.
With this patch:
Both channel drivers match incoming reason values with values documented
by REDIRECTING(reason) and values documented by RFC5806 regardless of
whether they are quoted or not. RFC5806 values are mapped to the
equivalent REDIRECTING(reason) documented value and is set in
REDIRECTING(reason). e.g., an incoming RFC5806 'unconditional' value or a
quoted string version ('"unconditional"') is converted to
REDIRECTING(reason)'s 'cfu' value. The user's dialplan only needs to deal
with 'cfu' instead of any of the aliases.
The incoming 480 response reason text supported by chan_sip checks for
known reason values and if not matched then puts quotes around the reason
string and assigns that to REDIRECTING(reason).
Both channel drivers send outgoing known REDIRECTING(reason) values as the
unquoted RFC5806 equivalent. User custom values are either sent as is or
with added quotes if SIP doesn't allow a character within the value as
part of a RFC3261 Section 25.1 token. Note that there are still
limitations on what characters can be put in a custom user value. e.g.,
embedding quotes in the middle of the reason string is silly and just
going to cause you grief.
* Setting a REDIRECTING(reason) value now recognizes RFC5806 aliases.
e.g., Setting REDIRECTING(reason) to 'unconditional' is converted to the
'cfu' value.
* Added missing malloc() NULL return check in res_pjsip_diversion.c
set_redirecting_reason().
* Fixed potential read from a stale pointer in res_pjsip_diversion.c
add_diversion_header(). The reason string needed to be copied into the
tdata memory pool to ensure that the string would always be available.
Otherwise, if the reason string returned by reason_code_to_str() was a
user's reason string then the string could be freed later by another
thread.
Change-Id: Ifba83d23a195a9f64d55b9c681d2e62476b68a87
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Change-Id: Id6350b3c7d4ec8df7ec89863566645e2b0f441fd
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* Fix double unref of other_party channel in off nominal path.
* This is unlikely to be a real problem. However, for safety,
in handle_incoming_request() keep the datastore ref with the
other_party channel ref until we are finished with the other_party
channel.
Change-Id: I78f22547bf0bb99fb20814ceab75952bd857f821
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This backs out item 4 of the 4875e5ac32f5ccad51add6a4216947bfb385245d
commit. Item 4 added the t38_bye_supplement. Unfortunately, the frame
that it puts into the bridge may or may not be processed by the time the
bridged peer is kicked out of the bridge. If it is processed then all is
well. However, if it is not processed then that channel is stuck in fax
mode until it hangs up or maybe if it joins another bridge for T.38
faxing.
ASTERISK-25582
Change-Id: Ib20a03ecadf1bf8a0dcadfadf6c2f2e60919a9f7
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There are a few cases where we're emitting notices or warnings
for things that really need neither, like a client retrying to subscribe
to mwi when they're not conifgured for it. They get a 404 so there's no
need for non-debug messages.
Change-Id: I05e38a7ff6c2f2521146f4be6a79731b9864e61f
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A few of the CLI commands weren't checking for enough arguments
and were SEGVing.
Change-Id: Ie6494132ad2fe54b4f014bcdc112a37c36a9b413
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Introduced realloaction of ast_str buf in sqlite3_escape functions in case
the returned buffer from threadstorage was actually too small.
Change-Id: I3c5eb43aaade93ee457943daddc651781954c445
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into 13
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log levels" into 13
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The 'reload' mechanism actually involves closing the underlying
socket and calling the appropriate udp, tcp or tls start functions
again. Only outbound_registration, pubsub and session needed work
to reset the transport before sending requests to insure that the
pjsip transport didn't get pulled out from under them.
In my testing, no calls were dropped when a transport was changed
for any of the 3 transport types even if ip addresses or ports were
changed. To be on the safe side however, a new transport option was
added (allow_reload) which defaults to 'no'. Unless it's explicitly
set to 'yes' for a transport, changes to that transport will be ignored
on a reload of res_pjsip. This should preserve the current behavior.
Change-Id: I5e759850e25958117d4c02f62ceb7244d7ec9edf
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