CLI command 'pjsip show contacts' inefficiently make a lot of DB requests.
For example if there are 10k aors then asterisk requests these 10k records
of aor and then does 10k requests of contact - one request per aor.
Even if use 'like <pattern>' the asterisk requests all aor's and contact's
records and then filters them by itself.
This patch gathers contact's container by
- retrieving all dynamic contacts by regex (filtered by reg_server)
- retrieving all aors with permanent contacts
- finally filters container by regex
ASTERISK-28077 #close
Change-Id: Id0ad65d14952a02fb213273a90f3f680a8149618
When REF_DEBUG and AO2_DEBUG are both enabled we closed the refs log
before we shutdown astobj2_container. This caused the AO2_DEBUG
container registration container to be reported as a leak.
Change-Id: If9111c4c21c68064b22c546d5d7a41fac430430e
Fixed an issue that resulted in "Allocation failed" each time an ARI
request was made to start playing MOH on a bridge.
In bridge_moh_create() we were attaching the after bridge callbacks to
chan which is the ;1 channel of the unreal channel pair. We should have
attached them to the ;2 channel which is pushed into the bridge by
ast_unreal_channel_push_to_bridge(). The callbacks are called when the
specific channel leaves the bridging system. Since the ;1 channel is
never put into a bridge the callbacks never get called. The callbacks
then never remove the moh_wrapper from the app_bridges_moh container. As
a result we cannot find the channel associated with the wrapper to start
MOH because it has hungup. This is the reason causing the reported issue.
* Rather than using after bridge callbacks to cleanup, we now have
moh_channel_thread() doing the cleanup when the channel hangs up.
* Fixed moh_channel_thread() accumulating control frames on the stasis
bridge MOH channel until MOH is stopped. Control frames are no longer
accumulated while MOH is playing.
* Fixed channel ref counting issue. stasis_app_bridge_moh_channel() may
or may not return a channel ref. As a result ast_ari_bridges_start_moh()
wouldn't know it may have a channel ref to release.
stasis_app_bridge_moh_channel() will now return a ref with the channel it
returns.
* Eliminated RAII_VAR in bridge_moh_create().
ASTERISK-26094 #close
Change-Id: Ibff479e167b3320c68aaabfada7e1d0ef7bd548c
This issue related to setting of holdtime, announcements, member delays.
It works well if we set the member delays to "0" and no announcements
and no holdtime.This issue will happen if we set member delays to "1",
"2"... or announcements or holdtime and hangs up the call during
processing it.
And here is the reason:
(At the step of answering a phone.)
It takes care any holdtime, announcements, member delays,
or other options after a call has been answered if it exists.
Normally, After the call has been aswered,
and we wait for the processing one of the cases of the member delays
or hold time or announcements finished, "if (ast_check_hangup(peer))"
will be not executed, then queue will be updated at update_queue().
Here, pending member will be removed.
However, after the call has been aswered,
if we hangs up the call during one of the cases of the member delays
or hold time or announcements, "if (ast_check_hangup(peer))"
will be executed.
outgoing = NULL and at hangupcalls, pending members will not be removed.
* This fixed patch will remove the pending member from container
before hanging up the call with outgoing is NULL.
ASTERISK-27920
Reported by: Cao Minh Hiep
Tested by: Cao Minh Hiep
Change-Id: Ib780fbf48ace9d2d8eaa1270b9d530a4fc14c855
This change raises a testsuite event to provide what port
Asterisk has actually allocated for RTP. This ensures that
testsuite tests can remove any assumption of ports and instead
use the actual port in use.
ASTERISK-28070
Change-Id: I91bd45782e84284e01c89acf4b2da352e14ae044
Use json_vsprintf from versions which contain fix for va_copy leak.
Apply fixes from jansson master:
* va_copy leak fix.
* Avoid potential invalid memory read in json_pack.
* Rename variable that shadowed another.
Change-Id: I7522e462d2a52f53010ffa1e7d705c666ec35539
When writing an RTCP report to json the code attempts to pack the "ssrc" and
"source_ssrc" unsigned integer values as a signed int value type. This of course
means if the ssrc's unsigned value is greater than that which can fit into a
signed integer value it gets converted to a negative number. Subsequently, the
negative value goes out in the json report.
This patch now packs the value as a json_int_t, which is the widest integer type
available on a given system. This should make it so the value no longer
overflows.
Note, this was caught by two failing tests hep/rtcp-receiver/ and
hep/rtcp-sender.
Change-Id: I2af275286ee5e795b79f0c3d450d9e4b28e958b0
The append_mailbox function wasn't calculating the correct length
to pass to ast_alloca and it wasn't handling the case where context
might be empty.
Found by the Address Sanitizer.
Change-Id: I7eb51c7bd18a7a8dbdba261462a95cc69e84f161
does_id_conflict() was passing a pointer to an int to a callback
that expected a pointer to a size_t.
Found by the Address Sanitizer.
Change-Id: I0ff542067eef63a14a60301654d65d34fe2ad503
On SQL error there is not diagnostic information about this error.
There is only
WARNING res_odbc.c: SQL Execute error -1!
The function ast_odbc_print_errors calls a SQLGetDiagField to get the number
of available diagnostic records, but the SQLGetDiagField returns 0.
However SQLGetDiagRec could return one diagnostic records in this case.
Looking at many example of getting diagnostics error information
I found out that the best way it's to use only SQLGetDiagRec
while it returns SQL_SUCCESS.
Also this patch adds calls of ast_odbc_print_errors on SQL_ERROR
to res_config_odbc.
ASTERISK-28065 #close
Change-Id: Iba5ae5470ac49ecd911dd084effbe9efac68ccc1
'rtpchecksums' and 'rtcpinterval' are not being reset to their defaults
if they are not present in the updated configuration file.
Change-Id: I1162e40199314d46cf3225d5e1271c4c81176670
app_voicemail wasn't properly cleaning up the stasis cache or the
mwi topic pool when the module was unloaded or when a user was
deleted as a result of a reload. This resulted in leaks in both
areas.
* app_voicemail now calls ast_delete_mwi_state_full when it frees
a user structure and ast_delete_mwi_state_full in turn now calls
the new stasis_topic_pool_delete_topic function to clear the topic
from the pool.
Change-Id: Ide23144a4a810e7e0faad5a8e988d15947965df8
The HTTP request processing in res_http_websocket allocates additional
space on the stack for various headers received during an Upgrade request.
An attacker could send a specially crafted request that causes this code
to overflow the stack, resulting in a crash.
* No longer allocate memory from the stack in a loop to parse the header
values. NOTE: There is a slight API change when using the passed in
strings as is. We now require the passed in strings to no longer have
leading or trailing whitespace. This isn't a problem as the only callers
have already done this before passing the strings to the affected
function.
ASTERISK-28013 #close
Change-Id: Ia564825a8a95e085fd17e658cb777fe1afa8091a
There's been a long standing leak when using topic pools. The
topics in the pool get cleaned up when the last pool reference is
released but you can't remove a topic specifically. If you reloaded
app_voicemail for instance, and mailboxes went away, their topics
were left in the pool.
* Added stasis_topic_pool_delete_topic() so modules can clean up
topics from pools.
* Registered the topic pool containers so it can be examined from
the CLI when AO2_DEBUG is enabled. They'll be named
"<topic_pool_name>-pool".
Change-Id: Ib7957951ee5c9b9b4482af7b9b4349112d62bc25
Since app_voicemail no longer uses the cache to maintain its state
there is no longer a need to cache these messages.
ASTERISK-27121
Change-Id: I321c708505f5ad8d00e1b0afc4c27dc2ac12ecb4
Stasis message types are global ao2 objects and we make stasis messages
and cache entries hold references to them. Since there are currently
situations where cache objects are never deleted, the reference count on
the types can exceed 100000 and generate a FRACK assertion message. The
stasis message cache could conceivably also have that many messages
legitimately on large systems.
The only down side to not holding the message type ref in the stasis
message is it only makes a crash either at shutdown or when manually
unloading a busy module slightly more likely. However, this is more
exposing a pre-existing stasis shutdown ordering issue than a problem with
not holding a message type ref in stasis messages.
* Made stasis messages and cache entries no longer hold a ref to the
message type.
Change-Id: Ibaa28efa8d8ad3836f0c65957192424c7f561707
* Create the stasis message object without a lock as it is immutable.
* Create the stasis message type object without a lock as it is immutable.
* Creating the stasis message type could crash if the passed in type name
is NULL and REF_DEBUG is enabled. Added missing NULL check when passing
the ao2 object tag string.
Change-Id: I28763c58bb9f0b427c11971d0103bf94055e7b32
In the original commit introducing the feature the column in the alembic
script was called 'suppress_q850_reason_header'.
In the code however the option is called 'suppress_q850_reason_headers'
(trailing 's'). This leads to errors when ARI push configuration is used.
Change-Id: Ie84808adbca6fcc9136556e4f5d741adbef5d14f
With tls and udp enabled asterisk generates a warning about sending
message via udp instead of tls.
sip notify command via cli works as expected and without warning.
asterisk has to set the connection information accordingly to connection
and not on presumption
ASTERISK-28057 #close
Change-Id: Ib43315aba1f2c14ba077b52d8c5b00be0006656e
app_voicemail was using the stasis cache to build and maintain a
list of mailboxes that had subscribers. It then used this list
to determine if a mailbox should be polled for new messages if
polling was enabled. For this to work, stasis had to cache every
subscription and unsubscription to the mailbox which caused a lot of
overhead, both cpu and memory related.
Since polling is only required when changes are being made to
mailboxes outside of app_voicemail and since the number of mailboxes
that don't have any subscribers is likely to be very low, all
mailboxes are now polled instead of just the ones with subscribers.
This paves the way for disabling the caching of stasis subscription
change messages.
Also fixed cleanup in some of the unit tests that not only left
test users in the users list but also caused segfaults if the tests
were run more than once.
ASTERISK-27121
Change-Id: I5cceb737246949f9782955c64425b8bd25a9e9ee
This change brings in PJSIP 2.8, removes all the patches
that were merged upstream, and makes a minor change to
support a breaking change that was done.
ASTERISK-28059
Change-Id: I5097772b11b0f95c3c1f52df6400158666f0a189
Both pjsip_tx_data.tp_info.dst_name and pjsip_rx_data.pkt_info.src_name
store IPv6 addresses without enclosing brackets. This causes some log
output to be confusing because it is difficult to separate the IPv6
address from a port specification.
* Use pj_sockaddr_print() along with pjsip_tx_data.tp_info.dst_addr and
pjsip_rx_data.pkt_info.src_addr where possible for consistent IPv6
output.
* When a pj_sockaddr is not available, explicitly wrap IPv6 addresses
in brackets.
* When assigning pjsip_rx_data.pkt_info.src_name ourselves, make sure
to also set pjsip_rx_data.pkt_info.src_addr.
Change-Id: I5cfe997ced7883862a12b9c7d8551d76ae02fcf8
Can't do anonymous http checkout from Security-testsuite.
Need to use same credentials as the gerrit review checkout.
Change-Id: I87af68c995cb8926f5e87f9af245600d76984f05
As they're not actively used, they only grow stale. The moduleinfo field itself
is kept in Asterisk 13/15 for ABI compatibility.
ASTERISK-28046 #close
Change-Id: I8df66a7007f807840414bb348511a8c14c05a9fc
