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Sometimes uw-imap function 'mail_fetchbody' returns huge len
which then pass to uw-imap function 'rfc822_base64'.
uw-imap tries to allocate huge memory and abort() on fail.
This patch check the len.
If the len more than max size (128 Mbytes) log error.
This patch also set variables len, newlen to avoid uninizialezed len.
This patch also check pointer returned by rfc822_base64.
ASTERISK-25899 #close
Change-Id: I4a0e7d655f11abef6a5224e2169df6d5c1f1caca
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Change-Id: Id1403b12136de62a272c01bb355aef65fd2c2d1e
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Because SQLite doesn't support full ALTER capabilities, alembic scripts
require batch operations. However, that capability wasn't available until
0.7.0 which some distributions haven't reached yet. Therefore, the batch
operations introduced in commit 86d6e44cc (review 2319) have been reverted
and SQLite is unsupported again, for now anyway.
Tested the full upgrade and downgrade on MySQL/Mariadb and Postgresql.
ASTERISK-25890 #close
Reported-by: Harley Peters
Change-Id: I82eba5456736320256f6775f5b0b40133f4d1c80
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When shutting down, the PJSIP sorcery is destroyed. The registrar
expiration module queries the PJSIP sorcery to determine what
to expire. As there was no synchronization between termination
of the expiration thread and the unloading of the module it was
possible for the thread to try to access the PJSIP sorcery after
it had been destroyed.
This change ensures that the thread is shut down before allowing
the module to be considered unloaded.
Change-Id: I69fd239edbaaf160c2d37ae00d3ac06e5596fe8b
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Due to a merge problem two options were swapped causing the
regcontext setting to not get set.
Change-Id: Icb33edc668e7357bacbaec2861a6b5ac64edaff1
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The problem is ast_frdup() does not copy whole frame.subclass for voice,
video and image frames, only the format is copied. For video frames, the
subclass structure contains the .frame_ending flag used to put the RTP
marker where it needs to be.
ASTERISK-25894 #close
Change-Id: I812ca90e84ed5d4f473b997d0dd0d3c5a915fe33
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Some SIP devices indicate hold/unhold using deferred SDP reinvites. In
other words, they provide no SDP in the reinvite.
A typical transaction that starts hold might look something like this:
* Device sends reinvite with no SDP
* Asterisk sends 200 OK with SDP indicating sendrecv on streams.
* Device sends ACK with SDP indicating sendonly on streams.
At this point, PJMedia's SDP negotiator saves Asterisk's local state as
being recvonly.
Now, when the device attempts to unhold, it again uses a deferred SDP
reinvite, so we end up doing the following:
* Device sends reinvite with no SDP
* Asterisk sends 200 OK with SDP indicating recvonly on streams
* Device sends ACK with SDP indicating sendonly on streams
The problem here is that Asterisk offered recvonly, and by RFC 3264's
rules, if an offer is recvonly, the answer has to be sendonly. The
result is that the device is not taken off hold.
What is supposed to happen is that Asterisk should indicate sendrecv in
the 200 OK that it sends. This way, the device has the freedom to
indicate sendrecv if it wants the stream taken off hold, or it can
continue to respond with sendonly if the purpose of the reinvite was
something else (like a session timer refresher).
The fix here is to alter the SDP negotiator's state when we receive a
reinvite with no SDP. If the negotiator's state is currently in the
recvonly or inactive state, then we alter our local state to be
sendrecv. This way, we allow the device to indicate the stream state as
desired.
ASTERISK-25854 #close
Reported by Robert McGilvray
Change-Id: I7615737276165eef3a593038413d936247dcc6ed
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In sorcery based config files where there are multiple categories with the same
name, you can't use the (+) operator to reliably append to a category because
config.c stops looking when it finds the first one with the same name.
Example:
[1000]
type = endpoint
[1000]
type = aor
[1000](+)
authenticate_qualify = yes
This config will fail because config.c appends authenticate_qualify to the
first category it finds, the endpoint, and that's not valid for endpoint.
Solution:
The capability to find a category that contains a certain variable already
exists so the only real change was to parse anything after the '+' that's not a
comma, as a filter string.
[1000]
type = endpoint
[1000]
type = aor
[1000](+type=aor)
authenticate_qualify = yes
This now works as expected.
Although the following example doesn't make any sense for pjsip, you can even
specify multiple filters:
[1000](+type=aor&qualify_frequency=10)
ASTERISK-25868 #close
Reported-by: Nick Repin
Change-Id: I10773da4c79db36fbf1993961992af63d3441580
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Websockets are a core part of ARI support and as such this
module should also be core supported.
Change-Id: I8f9283c6a167152761b92984779bb39e3db51a9c
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structures" into 13
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String fields are great, except that you can't add new ones without breaking
ABI compatibility because it shifts down everything else in the structure.
The only alternative is to add your own char * field to the end of the
structure and manage the memory yourself which isn't ideal, especially since
you then can't use the OPT_STRINGFIELD_T type.
Background:
The reason string fields had to be declared inside the
AST_DECLARE_STRING_FIELDS block was to facilitate iteration over all declared
fields for initialization, compare and copy. Since AST_DECLARE_STRING_FIELDS
declared the pool, then the fields, then the manager, you could use the offsets
of the pool and manager and iterate over the sequential addresses in between to
access the fields. The actual pool, field allocation and field set operations
don't actually care where the field is. It's just iteration over the fields
that was the problem.
Solution: Extended String Fields
An extended string field is one that is declared outside the
AST_DECLARE_STRING_FIELDS block but still (anywhere) inside the parent
structure. Other than using AST_STRING_FIELD_EXTENDED instead of
AST_STRING_FIELD, it looks the same as other string fields. It's storage comes
from the pool and it participates in string field compare and copy operations
peformed on the parent structure. It's also a valid target for the
OPT_STRINGFIELD_T aco option type.
Implementation:
To keep track of the extended fields and make sure that ABI isn't broken, the
existing embedded_pool pointer in the manager structure was repurposed to be a
pointer to a separate header structure that contains the embedded_pool pointer
plus a vector of fields. The length of the manager structure didn't change and
the embedded_pool pointer isn't used in the macros, only the stringfields C
code. A side benefit of this is that changing the header structure in the
future won't break ABI.
ast_string_fields_init initializes the normal string fields and appends them to
the vector, and subsequent calls to ast_string_field_init_extended initialize
and append the extended fields. Cleanup, ast_string_fields_cmp, and
ast_string_fields_copy can now work on the vector instead of sequentially
traversing the addresses between the pool and manager.
The total size of a structure using string fields didn't change, whether using
extended fields or not, nor have the offsets of any structure members, either
inside the original block or outside. Adding an extended field to the end of a
structure is the same as adding a char *.
Details:
The stringfield C code was pulled out from utils.c and into stringfields.c.
It just made sense.
Additional work was done in ast_string_field_init and
ast_calloc_with_stringfields to handle the allocation of the new header
structure and the vector, and the associated cleanup. In the process some
additional NULL pointer checking was added.
A lot of work was done in stringfields.h since the logic for compare and copy
is there. Documentation was added as well as somne additional NULL checking.
The ability to call ast_calloc_with_stringfields with a number of structures
greater than 1 never really worked. Well, the calloc worked but there was no
way to access the additional structures or clean them up. It was agreed that
there was no use case for requesting more than 1 structure so an ast_assert
was added to prevent it and the iteration code removed.
Testing:
The stringfield unit tests were updated to test both normal and extended
fields. Tests for ast_string_field_ptr_set_by_fields and
ast_calloc_with_stringfields were also added.
As an ABI test, 13 was compiled from git and the res_pjsip_* modules, except
res_pjsip itself, saved off. The patch was then added and a full compile and
install was performed. Then the older res_pjsip_* moduled were copied over the
installed versions so res_pjsip was new and the rest were old. No issues.
contact->aor, which is a char * at the end of contact, was then changed to an
extended string field and a recompile and reinstall was performed, again
leaving stock versions of the the res_pjsip_* modules. Again, no issues with
the res_pjsip_* modules using the old stringfield implementation and with
contact->aor as a char *, and res_pjsip itself using the new stringfield
implementation and contact->aor being an extended string field.
Finally, several existing string fields were converted to extended string
fields to test OPT_STRINGFIELD_T. Again, no issues.
Change-Id: I235db338c5b178f5a13b7946afbaa5d4a0f91d61
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16c7d8e74a9af13f98c3c22aa9c43ce39965f6b7" into 13
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I forgot the new voicemail_extension wasn't a stringfield and didn't check
for NULL where I should have.
Change-Id: I029482d5c2ab72474838750461bd46b0809c90fb
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into 13
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mwi_subscribe_replaces_unsolicited" into 13
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check_installed_debs wasn't handling virtual packages like libsrtp-dev and
libresample-dev and on multiarch systems it was accidentally filtering out all
packages if any :i386 packages were found instead of just filtering out the
:i386 packages themselves.
Change-Id: Ifd68da0d1ee30cc84df14de3f9b9079d7c3cecda
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ast_cli_allow_on_shutdown(e) should have been ast_cli_allow_at_shutdown(e).
Change-Id: I4f092495c0b2bfd85c2651e0b5877bf4d05d9faf
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Session-Timers." into 13
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creation" into 13
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into 13
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LDCONFIG apparently isn't set to something sane on all systems so the creation
of the shared library links fails. Instead of just testing for non-blank,
main/Makefile now checks that LDCONFIG is actually executable and reverts to
LN if it isn't.
This applies to both libasteriskpj and libasteriskssl.
Thanks to 'abelbeck' for pointing out that the issue was LDCONFIG.
ASTERISK-25873 #close
Reported-by: Hans van Eijsden
Change-Id: I25b76379bc637726ec044b2c0e709b56b3701729
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Change-Id: Ifadc469590bd4d5368e19d3763db3bd1f80fdb95
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The only caller of ari_bridges_play_found() has this note:
If ari_bridges_play_found fails because the channel is unavailable for
playback, The channel will be removed from the playback list soon. We can
keep trying to get channels from the list until we either get one that
will work or else there isn't a channel for this bridge anymore, in which
case we'll revert to ari_bridges_play_new.
Change-Id: Ib068141b367ccaa17be0dab4181c98e26c5127d6
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Change-Id: Ic7d93c402c498677a122505558859c853d4e5ac7
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Change-Id: I0be0627260cd8d6b6c3cc345949dcfdf32eff1f3
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The stasis_app_playback and stasis_app_recording structs need to have a
struct stasis_app_control ref. Other threads can get a reference to the
playback and recording structs from their respective global container.
These other threads can then use the control pointer they contain after
the control struct has gone.
* Add control ref to stasis_app_playback and stasis_app_recording structs.
With the refs added, the control command queue can now have a circular
control reference which will cause the control struct to never get
released if the control's command queue is not flushed when the channel
leaves the Stasis application. Also the command queue needs better
protection from adding commands if the control->is_done flag is set.
* Flush the control command queue on exit.
ASTERISK-25882 #close
Change-Id: I3cf1fb59cbe6f50f20d9e35a2c07ac07d7f4320d
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* Give the struct stasis_app_control ao2 object a ref to the channel held
in the object. Now the channel will still be around if a thread needs to
post a stasis message instead of crash because the topic was destroyed.
* Moved stopping any lingering silence generator out of the struct
stasis_app_control destructor and made it a part of exiting the Stasis
application. Who knows which thread the destructor will be called under
so it cannot affect the channel's silence generator. Not only was the
channel unprotected when the silence generator was stopped, stasis may no
longer even control the channel.
ASTERISK-25882
Change-Id: I21728161b5fe638cef7976fa36a605043a7497e4
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Background:
If your extension is 1000 and the voicemail access extension is 1571 and you
dial 1571, usually a dialplan rule calls voicemailmain with your extension and
you are placed directly in your mailbox. Therefore most admins program the
voicemail (or other speed dial) button on their phones to the access extension.
Some phones (Snom at least) use whatever is programmed there to also subscribe
for MWI and so can't dial one number and subscribe to another. This works fine
in chan_sip because chan_sip completely ignores the user portion of the
SUBSCRIBE message request URI. If it can match the peer, is subscribes to the
peer's mailbox. The user could be set to anything or nothing and you'd still
get subscribed to your mailbox.
Issue:
chan_pjsip actually uses the user portion of the URI to find an aor and its
mailboxes. Therefore a subscribe to 1571 results in a 404. Sure, you can
create an aor for 1571 but you certainly can't add your entire voicemail
system's mailboxes to it and everyone would get notified of every MWI.
Solution:
When an MWI subscribe comes in and an aor can't be found that matches the
resource directly, check the resource against the endpoint's aors. If an aor
is found that has a voicemail_extension that matches the resource, use it.
ASTERISK-25865
Reported-by: Ross Beer
Change-Id: I770ea185f751f1ada888fafb4b452115f1c06e9e
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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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into 13
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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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When using app_echo via WebRTC with VP8 video the video would appear
only after a few minutes, because there would be nothing to request
a full reference frame.
This fixes the problem in both ways:
- echos any VIDUPDATE frames received on the channel
- sends one such frame when first video frame is to be forwarded
This makes the echo work with Firefox and Chrome WebRTC implementation.
ASTERISK-25867 #close
Change-Id: I73bda87bf7532ee8bfb28d917045a21034908c1e
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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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