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Added functions that convert a string to an unsigned integer or unsigned long.
A couple of unit test were also created to test the routines. The reasons for
adding these conversion utilities (and hopefully eventually more) are as
follows:
* Conversion routines are functionally contained with consistent and
better error checking
* The function names offer a better description of what is happening
* It encourages code reuse for easier bug fixing at a single source
* It's simpler to use
* It's unit testable
For instance, currently in a lot of places when converting to an integer or
similar the "sscanf" function is used. When using "sscanf" it may not be
immediately clear what's happening as it lacks semantic naming. Limited error
checking is usually done as well. For example, most of the time a check is done
to make sure the value converted, but does not check for overflows or negative
valued conversions when converting unsigned numbers.
Why use/wrap "strtoul" and not "sscanf" then? Primarily, it lacks some of the
built in error handling that "strtoul" has. For instance "strtoul" contains
overflow checks. Less so, but can still factor as reasons, "sscanf" is slightly
more complex in its use. And maybe a bit controversial, but it may be ("big if")
potentially slower than "strtoul" in some cases.
Change-Id: If7eaca4a48f8c7b89cc8b5a1f4bed2852fca82bb
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Retransmissions of an initial INVITE could be queued in the serializer
before we have processed the first INVITE message. If the first INVITE
message doesn't get completely processed before the retransmissions are
seen then we could try to setup the same call from the retransmissions. A
symptom of this is seeing a (key exists) message associated with an
INVITE. An earlier change attempted to address this kind of problem by
calculating a distributor serializer to use for unassociated messages.
Part of that change also made incoming calls keep using that distributor
serializer. (ASTERISK-26088) However, some leftover code was still
deferring the INVITE processing to the session's serializer even though we
were already in that serializer. This not only is unnecessary but would
cause the same call resetup problem.
* Removed the code to defer processing the initial INVITE to the session's
serializer because we are already running in that serializer.
ASTERISK-26998 #close
Change-Id: I1e822d82dcc650e508bc2d40d545d5de4f3421f6
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Change-Id: Ie1c2d83af66f27a449da09a68d987e0992627fee
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* changes:
SDP: Make process possible multiple fmtp attributes per rtpmap.
SDP: Explicitly stop a RTP instance before destoying it.
SDP: Rework merge_capabilities().
SDP: Update ast_get_topology_from_sdp() to keep RTP map.
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Return cahnnel nativeformats to fix bridge technology selection process.
Same approach as in pjsip module.
ASTERISK-26143
Reported-by: Henning Holtschneider
Change-Id: I64e863753954d6ad67a9e722df2ebc328705ad48
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Change-Id: Ie7511008d82b59590e0eb520a21b5e1da4bd7349
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The sdp_state.remote_capabilities was only used inside merge_sdps() and
subsequent calls to merge_sdps() by re-INVITE's would leak them.
Change-Id: I0ceb7838ea044cc913e8ad4a255c39c9740ae0ce
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When we optionally set the interface_address we are forcing the media to
go out a specific interface address. This allows us to optionally have
the media go out the interface that SIP signalling came in on or if we are
configured to have the media always go out a specific address.
Change-Id: I160d9fac322a075bd2557b430632544178196189
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* Made sdp_add_m_from_rtp_stream() and sdp_add_m_from_udptl_stream()
handle generating disabled/declined streams.
* Added /main/sdp/sdp_merge_asymmetric unit test. It currently does not
check the offerer side negotiated SDP because that isn't the purpose of
this patch and there is much to be done to handle declined/dummy streams.
* Added T.38 image streams to the /main/sdp/sdp_merge_symmetric and
/main/sdp/sdp_merge_crisscross unit tests.
Change-Id: Ib4dcb3ca4f9a9133b376f4e3302f9a1f963f2b31
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* Tried to give better variable names.
* Made our SDP answer use the offer's RTP payload types as the SDP RFC
says we SHOULD.
* Updating the local topology now takes the stream format caps. We are
likely preparing to send an offer.
Change-Id: I34d3be8e3036402a8575ffcae3eebc5ce348d7c0
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* Add failure exits to ast_get_topology_from_sdp().
Change-Id: I4cc85c1ede8d712766ed20f544dbcef04c8c1049
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This change uses the functions provided by OpenSSL to query
and better construct error messages for situations where
the connection encounters a problem.
ASTERISK-26606
Change-Id: I7ae40ce88c0dc4e185c4df1ceb3a6ccc198f075b
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It is possible to initialize a valid config without a capath
or cafile definition. This will cause a crash on a reload.
This fix ensures capath is always allocated.
ASTERISK-26983 #close
Change-Id: I63ff715d9d9023427543a5b8a4ba7b0d82533c12
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When a column is of type timestamp, the fraction part of the event
field's seconds was frequently parsed incorrectly especially if
there were leading zeros. For instance "2017-05-23 23:55:03.023"
would be parsed into an int as "23" then when the timestamp was
formatted again to be inserted into the database column it'd be
"2017-05-23 23:55:03.23" which is now 230 milliseconds instead of
23 milliseconds. "03.000001" would be transformed to "03.1", etc.
* If the event field is 'eventtime' and the db column is timestamp,
then existing processing has already correctly formatted the
timestamp so now we simply use it rather than parsing it and
re-printing it. This is the most common use case anyway.
* If the event field is other than 'eventtime' and the db column
is timestamp, we now parse the seconds, including the fractional
part into a double rather than 2 ints. This preserves the
magnitude and precision of the fractional part. When we print
it, we now print it as a "%09.6lf" which correctly represents the
input.
To be honest, why we parse the string timestamp into components,
test the components, then print the components back into a string
timestamp is beyond me. We should use parse it, test it, then if
it passes, use the original string representation in the database
call. Maybe someone thought that some implementations wouldn't
take a partial timestamp string like "2017-05-06" and decided to
always produce a full timestamp string even if an abbreviated one
was supplied. Anyway, I'm leaving it as it is.
ASTERISK-25032 #close
Reported-by: Etienne Lessard
Change-Id: Id407e6221f79a5c1120e1a70bc7e893bbcaf1938
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This change adds the required logic to allow the SIP
Call-ID to be placed into the HEP RTCP traffic if the
chan_sip module is used. In cases where the option is
enabled but the channel is not either SIP or PJSIP then
the code will fallback to the channel name as done
previously.
Based on the change on Nir's branch at:
team/nirs/hep-chan-sip-support
ASTERISK-26427
Change-Id: I09ffa5f6e2fdfd99ee999650ba4e0a7aad6dc40d
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All log messages go to a queue serviced by a single thread
which does all the IO. This setting controls how big that
queue can get (and therefore how much memory is allocated)
before new messages are discarded. The default is 1000.
Should something go bezerk and log tons of messages in a tight
loop, this will prevent memory escalation.
When the limit is reached, a WARNING is logged to that effect
and messages are discarded until the queue is empty again. At
that time another WARNING will be logged with the count of
discarded messages. There's no "low water mark" for this queue
because the logger thread empties the entire queue and processes it
in 1 batch before going back and waiting on the queue again.
Implementing a low water mark would mean additional locking as
the thread processes each message and it's not worth it.
A "test" was added to test_logger.c but since the outcome is
non-deterministic, it's really just a cli command, not a unit
test.
Change-Id: Ib4520c95e1ca5325dbf584c7989ce391649836d1
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ast_stream_clone() cannot copy the opaque user data stored on a stream.
We don't know how to clone the data so it isn't copied into the clone.
Change-Id: Ia51321bf38ecbfdcc53787ca77ea5fd2cabdf367
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Change-Id: I532052bd7cd95a4b3565485fc01e2a1ea07ee647
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A regression was introduced in 12 where passing an empty value
to the CDR dialplan function was not longer allowed. This
change returns to the behavior of 11 where it is permitted.
ASTERISK-26173
Change-Id: I3f148203b54ec088007e29e30005a5de122e51c5
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menu_template_handler wasn't properly accounting for the fact that
it might be called both during a load/reload (which isn't really
valid but not prevented) and by a dialplan function. In both cases
it was attempting to use the "pending" config which wasn't valid in
the latter case. aco_process_config is also partly to blame because
it wasn't properly cleaning "pending" up when a reload was done and
no changes were made. Both of these contributed to a crash if
CONFBRIDGE(menu,template) was called in a dialplan after a reload.
* aco_process_config now sets info->internal->pending to NULL
after it unrefs it although this isn't strictly necessary in the
context of this fix.
* menu_template_handler now uses the "current" config and silently
ignores any attempt to be called as a result of someone uses the
"template" parameter in the conf file.
Luckily there's no other place in the codebase where
aco_pending_config is used outside of aco_process_config.
ASTERISK-25506 #close
Reported-by: Frederic LE FOLL
Change-Id: Ib349a17d3d088f092480b19addd7122fcaac21a7
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When using the Bridge AMI action on the same channel multiple times
it was possible for the channel to return to the wrong location in
the dialplan if the other party hung up. This happened because the
priority of the channel was not preserved across each action
invocation and it would fail to move on to the next priority in
other cases.
This change makes it so that the priority of a channel is preserved
when taking control of it from another thread and it is incremented
as appropriate such that the priority reflects where the channel
should next be executed in the dialplan, not where it may or may not
currently be.
The Bridge AMI action was also changed to ensure that it too
starts the channels at the next location in the dialplan.
ASTERISK-24529
Change-Id: I52406669cf64208aef7252a65b63ade31fbf7a5a
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This patch is the first cut at adding stream support to the bridging framework.
Changes were made to the framework that allows mapping of stream topologies to
a bridge's supported media types.
The first channel to enter a bridge initially defines the media types for a
bridge (i.e. a one to one mapping is created between the bridge and the first
channel). Subsequently added channels merge their media types into the bridge's
adding to it when necessary. This allows channels with different sized
topologies to map correctly to each other according to media type. The bridge
drops any frame that does not have a matching index into a given write stream.
For now though, bridge_simple will align its two channels according to size or
first to join. Once both channels join the bridge the one with the most streams
will indicate to the other channel to update its streams to be the same as that
of the other. If both channels have the same number of streams then the first
channel to join is chosen as the stream base.
A topology change source was also added to a channel when a stream toplogy
change request is made. This allows subsystems to know whether or not they
initiated a change request. Thus avoiding potential recursive situations.
ASTERISK-26966 #close
Change-Id: I1eb5987921dd80c3cdcf52accc136393ca2d4163
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When a call gets put on hold RTP is temporarily stopped and Asterisk was
setting the remote RTCP address to NULL. Then when RTCP data was received
from the remote endpoint, Asterisk would be missing this information when
publishing the rtcp_message stasis event. Consequently, message subscribers
(in this case res_hep_rtcp) trying to parse the "from" field output the
following error:
"ast_sockaddr_split_hostport: Port missing in (null)"
This patch makes it so the remote RTCP address is no longer set to NULL when
stopping RTP. There was only one place that appeared to check if the remote
RTCP address was NULL as a way to tell if RTCP was running. This patch added
an additional check on the RTCP schedid for that case to make sure RTCP was
truly not running.
ASTERISK-26860 #close
Change-Id: I6be200fb20db647e48b5138ea4b81dfa7962974b
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connections"
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Many sound files don't have a full frame's worth of data at EOF, so the
warning messages were a bit too noisy. So we demote them to debug
messages.
Change-Id: I6b617467d687658adca39170a81797a11cc766f6
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The telephone_event option was used as a flag and a bit mapped value in
different places when it is a boolean. It is also inadequate to configure
the DTMF operation of the RTP instance created for the stream.
Change-Id: Ib1addeaf0ce86f07039f2f979cab29405dc5239b
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cap."
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A deadlock can happen between a channel lock and a pjsip session media
container lock. One thread is processing a reINVITE's SDP and walking
through the session's media container when it waits for the channel lock
to put the determined format capabilities onto the channel. The other
thread is writing a frame to the channel and processing the T.38 frame
hook. The T.38 frame hook then waits for the pjsip session's media
container lock. The two threads are now deadlocked.
* Made the T.38 frame hook release the channel lock before searching the
session's media container. This fix has been done to several other
frame hooks to fix deadlocks.
ASTERISK-26974 #close
Change-Id: Ie984a76ce00bef6ec9aa239010e51e8dd74c8186
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