get an EOP, we no longer worry whether the final stages tidy up. This helps
tolerate the increasing number of VoIP calls which hang up before the audio has
flushed down the line.
A few little cleanups.
Use peerhangup flag variable to track whether a hangup has been
initiated by the peer or libpri itself (e.g. Layer 2 timeouts).
These changes fix a couple of problems with hangup events not being
handled properly in some situations:
- Call abort caused by incoming RESTART on a channel in use
- T309 timeout after L2 loss
- Improved hangup handling in libpri-side on_hangup() event handler
and state_advance() (FreeTDM side)
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
PROGRESS without PROGRESS INDICATOR IE is invalid according to Q.931,
so always call pri_progress() with info flag set, even if we do not
have media yet.
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
"Safer" version of ftdm_clamp(), that swaps min/max parameters if
vmin happens to be larger than vmax, making sure the output will
always satisfy vmin <= x <= vmax.
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
The default remains the same: we show the huge ClueCon nag banner on
startup and in fs_cli.
However, if you pass --disable-huge-cluecon-nag, no banner will be
shown.
If you pass --enable-modest-cluecon-nag, a modest text-based ClueCon
reminder will be shown instead.
Some people, it seems, were having sleepless nights from the
log messages this feature produces every 15 minutes.
Default disable the feature, because i hate repeating myself
over and over, explaining them this is a non-issue and intended
behaviour and how to disable it in the config.
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
FTDM_SIGEVENT_UP was falling through to FTDM_SIGEVENT_PROGRESS_MEDIA, which
does (almost) the same, so no harm, but still fix it.
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
Channels can be in DOWN state but reserved by an incoming call
(FTDM_CHANNEL_INUSE flag). Additionally check the flag in on_timeout_t3xx()
and skip those channels.
Reported-by: privi #freetdm @ irc.freenode.net
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
Do not try to send RESTART on BRI PTMP spans, libpri will just print
an error and do nothing, causing the channels to be stuck in RESTART
forever.
Add T316 (RESTART ACK timeout), which is not implemented by libpri.
The default timeout is 30 seconds (Q.931 recommends 2 minutes, but
that is a little long and libpri layer 2 is being stupid) and the
restart attempt limit to 3 (instead of 2).
Periodically send RESTART on idle b-channels in PTP NT mode.
Default interval is 15 minutes, feature can be disabled by setting
"idle_restart_interval" to 0 in the span configuration.
Allow timeout / interval values to have an additional time unit
suffix for convenience reasons, the default (no unit specified)
is milliseconds, valid units include:
w (week), d (day), h (hour), m (minute), s (second)
Only full integers are accepted, no fractional numbers, valid examples:
2w = two weeks
37m = 37 minutes
1h = 1 hour
Combinations of multiple numbers and units (e.g. "1w5d") are not supported.
New span configuration parameters:
idle_restart_interval (milliseconds / time unit suffix /
0 = disabled)
t316 / restart_timeout (milliseconds / time unit suffix)
t316_limit / restart_attempts (number of max. attempts)
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
Add per b-channel private data (ftdm_libpri_b_data_t), move the q.931 libpri
call pointer into it and use channel->call_data to make it accessible.
Use ftdm_channel_wait() instead of select() to wait for I/O events on
the D-Channel and rewrite the timeout handling, adding support for custom
timers that can be added to the b-channel private data and started/stopped
with lpwrap_timer_start()/_stop().
The (sorted) list of active lpwrap timers is protected by the timer_mutex,
which is initialized in the now unified lpwrap_init_pri() (see below) and
destroyed in the new lpwrap_destroy_pri() function.
Unify lpwrap_init_pri() and lpwrap_init_bri(). Use the span type to select
BRI/PRI and Point-to-Point vs. Multipoint settings.
Use the new custom lpwrap timer API to implement T302 (aka overlap receiving
digit timeout). The default T302 timeout is 3000 miliseconds and can be set
via the "digit_timeout" or "t302" span configuration parameter
(0 = disable timer, valid range = 3000 ... 30000 miliseconds).
The lpwrap_pri_init() changes made it neccessarry to rewrite the span runtime
thread event loop. The d-channel initialization and libpri event handler
registration have been moved out of the event loop. The b-channel restart
handling in the event loop has been left as-is, pending a further clean up in
the future.
Rename on_info() to on_information() and add a new on_keypad_digit() event
handler to convert incoming KEYPAD DIGIT messages to channel DTMF events
(only useful on BRI PTMP NT-mode).
The channel restart handling has been improved to distinguish between locally
triggered and restarts caused by the remote end (= incoming RESTART messages).
Restarts on the D-Channel now use pri_restart(), causing a Q.921 reset on
the span.
Large patch, i'd have loved to split this into smaller pieces and actually
test some of the bits more toroughly (restart handling), but some people
are f*cking annoying, so here it is.
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
AC_PROG_LIBTOOL
m4_include(m4/acinclude.m4)
Adding these and taking out all the Makefile.in files from the tiff-4.0.2
directory seems to sort out the build issues.
Move the B-channel message handling into a per-span I/O thread,
to solve most of the problems caused by the intermixed data + control
socket interface of mISDN, missing write poll() support on
mISDN B-channels and the FreeTDM I/O model. This eliminates most of
the audio problems (except for a few minor glitches).
A unix stream socket pair is used as a bi-directional pipe replacement
(the pipe code is still included in this commit, but will be removed later),
with the RX and TX buffer sizes carefully tuned to avoid excessive buffering
(= latency) and a deadlock situation between the write() call in ftdm_write()
and the code in misdn_span_run() that needs a minimum amount of data in the
TX buffer, before sending out a PH_DATA_REQ to the mISDN socket
(see misdn_span_run() comments for more details).
The minimum size for pipes is PAGE_SIZE (4k), which is ~500 ms worth of
audio. A socket pair RX/TX buffer size of 3k, seems to hold a maximum
amount of around 500 bytes data in practice, giving us a much lower
maximum latency than a unix pipe. (The socket pair might be replaced by a
custom ring buffer / fifo data structure to get even more fine grained
control of the maximum latency.)
The newly introduced span_start / span_stop callbacks in
ftdm_io_interface_t are used to start / stop the I/O thread. The callback
functions will wait up to 10 seconds for the thread to successfully
start up or shut down (using a mutex + condition var).
NOTE: Using any of the locking ftdm_span_() functions in the I/O will cause
a deadlock between the I/O thread (trying to lock span->mutex) and the
thread calling ftdm_start()/_stop() (holding the span->mutex).
(The I/O thread currently uses direct span member access to avoid this.)
The I/O thread uses the epoll(7) family of functions for event handling.
An epoll context is created on startup and all B-channel sockets are
registered (READ, PRI and ERR). Before entering the event loop,
the I/O thread will send a signal on the condition variable, to
indicate it has completed the startup procedure.
Incoming b-channel and command pipe events are handled by the event loop.
Payload of incoming PH_DATA_IND frames (= audio data) is sent to the
rx_audio_pipe_in end of the b-channel's socket pair and, if enough data is
available, a PH_DATA_REQ of the same size is sent to the b-channel mISDN socket
to transmit audio.
A MISDN_CMD_STOP command on the event pipe will wake up the I/O thread and
cause it to shut down. All b-channels will be unregistered from the epoll context
and the epoll fd closed. The I/O thread terminates itself after signalling the
successfull shutdown on the condition variable.
TODOs:
- Move D-Channel into I/O thread too
- Custom FIFO/ring buffer for data (even lower latency)
- Improve epoll() code (per-channel struct w/ callback, for epfd.data.ptr)
- Use mISDN DSP for audio (e.g. tone generator, dtmf detector, echo cancel)
- Use a per-port / span control socket to execute channel commands
synchronously, or add misdn_commands (queue?) that can be used that way
- Name I/O threads 'mISDN-%SPAN_NAME%', e.g. 'mISDN-M_BRI1'
(= add ftdm_thread_set_namef(thread, fmt, ...) / ftdm_thread_set_name(thread, name))
TL;DR: "tweak", solves "booboo" with audio
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
Hunt for a free channel for incoming calls that do not
preselect a channel (pevent->ring.channel == -1).
Verify the preselected channel for calls that do specify a channel
and in case the channel is already taken, hunt for a free one,
or abort with an error message (if the preselection was exclusive).
TE-mode channel selection is the same as before
(there's still room for improvement, though, but i'll save that for later).
The MSN/DDI filter code is moved into the TE-mode section (only useful there).
The duplicate ring detection had to be reworked. We now store the
call reference (CRV) in caller_data->call_reference of the selected channel
and do a CRV -> channel look up with find_channel_by_cref()
at the top of on_ring().
NOTE: This is only lightly tested (NT/TE mode), i'd either have to
write a lot of custom code to check it toroughly or the need for
a scriptable ISDN stack...
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
Enhancements to trace logging, include threads and context ID.
Changed default opal_conf.xml to allow more than just G.711 uLaw and not to clutter log file with debug logs.
Added to opal_conf.xml item for "disable-transcoding".
Updated build/buildopal.sh to use correct ./configure items for PTLib, allow for something other than standard install directory for PTLib/OPAL and be able to easily bind to a specific release of PTLib/OPAL.
Callbacks are invoked from ftdm_span_start/_stop().
I/O is started before SIG and shut down in reverse order.
This is needed for ftmod_misdn, to move the mISDN message handling
into a separate thread (solving the mISDN socket vs. FreeTDM API issues).
With these callbacks, the I/O thread can be started after the span I/O configuration
has been (successfully) completed and stopped before destroying the span.
NOTE: Both SIG and I/O callbacks are called with the span mutex locked,
so threads created or destroyed synchronously in either of the custom
start/stop functions, can not use ftdm_span_*() functions that lock
the span mutex (e.g. ftdm_span_get_channel_count()).
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
The --enable-builtin-tiff option appends libs/tiff-3.8.2/libtiff/libtiff.la
to LIBS, causing the AC_CHECK_LIB([jpeg]...) check to fail, because
libtiff.la does not exist at configure time.
Temporarily store tiff and jpeg libs in TIFF_-/JPEG_LIBS variables and
append them to LIBS after all library checks have run.
Example error output:
configure:20049: checking for jpeg_start_compress in -ljpeg
configure:20074: cc -o conftest -O2 -pipe -fno-strict-aliasing -L/usr/local/lib conftest.c -ljpeg -lm /usr/home/ports/net/freeswitch-core-devel/work/freeswitch-1.2.1/libs/tiff-3.8.2/libtiff/libtiff.la >&5
cc: /usr/home/ports/net/freeswitch-core-devel/work/freeswitch-1.2.1/libs/tiff-3.8.2/libtiff/libtiff.la: No such file or directory
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
The len variable can, in certain situations (large burst of incoming non-SLIN audio),
exceed the size of the on-stack frame buffer, which causes ftdm_buffer_read_loop() to
overwrite the dt_buffer pointer.
Use ftdm_min() to make sure len (after conversion to SLIN units) isn't larger
than the frame buffer size.
Also adds are couple more code comments.
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
Resolves OPENZAP-189
Fixes:
src/ftmod/ftmod_isdn/ftmod_isdn.c: In function 'ftdm_isdn_931_34':
src/ftmod/ftmod_isdn/ftmod_isdn.c:902:21: error: variable 'status' set but not used [-Werror=unused-but-set-variable]
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
. added called party number INN variable
ie. <action application="export" data="freetdm_ss7_cld_inn=0"/>
. added multiple variables to set User Service Information IE
variables are :
ss7_iam_usi_trans_cap
ss7_iam_usi_code_standard
ss7_iam_usi_trans_mode
ss7_iam_usi_trans_rate_0
ss7_iam_usi_trans_rate_1
ss7_iam_usi_layer1_ident
ss7_iam_usi_layer1_prot
ss7_iam_usi_layer2_ident
ss7_iam_usi_layer2_prot
ss7_iam_usi_layer3_ident
ss7_iam_usi_layer3_prot
ss7_iam_usi_chan_struct
ss7_iam_usi_config
ss7_iam_usi_establish
ss7_iam_usi_symmetry
ss7_iam_usi_rate_multiplier
. ss7_iam_usi_trans_cap is a string variable, the others are all integers
. ss7_iam_usi_trans_cap has options of :
- SPEECH
- UNRESTRICTED
- RESTRICTED
- 31KHZ
- 7KHZ
- 15KHZ
- VIDEO
default value is SPEECH, if the parameter is wrong, set to SPEECH. If not
set, this field is not present
Fixes:
src/ftmod/ftmod_misdn/ftmod_misdn.c: In function 'misdn_handle_mph_information_ind':
src/ftmod/ftmod_misdn/ftmod_misdn.c:871:3: error: format '%lx' expects argument of type 'long unsigned int', but argument 13 has type 'uint64_t' [-Werror=format]
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
Do a "soft" wraparound with modulo, removes the ~0.5s tone glitch.
(Multiply ts.rate (samples) by two to match the offset unit (bytes, 2 per sample).)
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
Store the offset in the teletone buffer in the b-channel private data.
An NT-mode setup with ftmod_misdn showed severe (dial-)tone distortions
in a sound editor (330Hz sine wave phase errors), caused by
using a global teletone buffer offset.
Switching to a per-channel offset, that is advanced by the amount
of data actually written to the channel, removes (almost) all
distortions.
There is still a minimal phase error every ~500ms (audible) that
needs more investigating.
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
Might as well import these too...
ftdm_offset_of() - Get offset of member in structure.
ftdm_container_of() - Get pointer to enclosing structure from pointer to structure member.
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
ftdm_min(x,y) - Returns the smaller of the two values x and y.
ftdm_max(x,y) - Returns the larger of the two values x and y.
ftdm_clamp(val, min, max) - Returns value that is in the range [vmin,vmax].
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
Remove the 'state' variable of per-channel data, use active flag exclusively to track
open/close state.
Add misdn_activate_channel()/misdn_deactivate_channel() helper functions, rename old
one to _misdn_toggle_channel() (internal).
Add _nowait variant of channel de-/activation function, that just sends the mISDN request message.
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
Store call CRV in caller_data.call_reference and use call private to hold the
FreeTDM channel object.
Remove isdn_data->channels_{local,remote,outbound}_crv arrays.
Allow (and force) inbound call state transition DIALTONE -> DOWN for
incoming RELEASE COMPLETE messages in NT mode.
Dialtone in NT mode works, everything else needs more testing.
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
ftmod_misdn currently returns len == 0 if the incoming message,
that triggered the read() call, does not contain any data.
Users of ftdm_channel_read() need to handle this case, or they
may possibly end up in an endless loop.
This patch reworks the ftdm_channel_read() handling in ftmod_isdn
and prevents it from entering an endless loop. The read error counter
is reset on first sucessful read w/ data.
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
cc1: warnings being treated as errors
./src/ftmod/ftmod_libpri/ftmod_libpri.c: In function 'msn_filter_foreach':
./src/ftmod/ftmod_libpri/ftmod_libpri.c:422: warning: dereferencing type-punned pointer will break strict-aliasing rules
make: *** [ftmod_libpri_la-ftmod_libpri.lo] Error 1
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
Fixes:
cp ESL.so /var/tmp/por.../image/usr/lib64/php5.3/lib/extensions/no-debug-zts-20090626
cp: cannot create regular file `/var/tmp/por.../image/usr/lib64/php5.3/lib/extensions/no-debug-zts-20090626': No such file or directory
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
Two fixes:
Use ftdm_set_string() instead of sprintf() (seriously, wtf?).
Drop invalid, needless argument to ftdm_log().
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
Neither of them accepts extra arguments and ftmod_sangoma_isdn is calling
ftmod_assert() with a char * msg argument.
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
The format string checks already caught a couple crash-worthy bugs and this
commit fixes a couple more.
Also includes __ftdm_check_scanf(), for completeness (currently unused).
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
Output the current trunk_type in "add X-channel vs. trunk_type" error messages and
check this for B-/D-channels too.
ISDN (= digital) spans need to have a trunk_type set before adding channels,
bail out early with an error message (actually two) if this is not the case.
(Adding channels should really be moved out of the parsing loop, to catch
certain types of errors easier.)
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
Some I/O implementations (e.g. mISDN) need this information to correctly
set up the port configuration. In FreeTDM, the mode of a span has been.
up until now, a part of the signalling module configuration ("node" or "mode"
for libpri and isdn spans), which is parsed _after_ the I/O part of a span
has been initialized. This limitation currently prevents us from using mISDN
in NT mode.
To work around this problem (without adding a lot of new tunk_type values like "E1_NET" etc.),
add a new "trunk_mode" parameter to the span categories in freetdm.conf, which by default
is "CPE" for all trunk types, except "FXS" (which defaults to "NET").
ftmod_misdn uses trunk_type + trunk_mode to correctly choose the d-channel protocol
for a port.
ftmod_libpri uses trunk_mode as a hint for the default "mode" setting.
(NOTE: It will print a warning if trunk_mode and signalling mode do not match.)
All other modules currently ignore the value of trunk_mode.
Example freetdm.conf snippet for a mISDN span in NET/NT mode
(using a HFC-S USB dongle):
[span misdn MBRI_1]
trunk_type => BRI_PTMP
trunk_mode => NET
b-channel => HFC-S_USB.1:1-2
d-channel => HFC-S_USB.1:3
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
Portability fix for uClibc and other (linux) environments that lack execinfo.h.
ftdm_backtrace_walk() and related return FTDM_NOTIMPL and print a message
if backtraces are not available in the current environment.
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
Indicator in IAM message.
- ss7_iam_nature_connection_hex is the channel variable to carry
HEX value from/to IAM message
- X-FreeTDM-IAM-NATURE-CONN-HEX is the x-header string to carry
HEX value from/to sip invite message
- This IE is 8 bit hex, which turns into 2 chars
- Only Satellite indicator, Continuity check indicator, and Echo
control device indicator are supported. Spare value are not taken.
Spare value is always set to all zero.
- example:
To set 00010101 (bit H-A), set X-FreeTDM-IAM-NATURE-CONN-HEX string
value to 15 in the sip invite message to NSG.
When NSG send out sip invite, if the incoming IAM message comes with
Nature of Connection IE, NSG will put X-FreeTDM-IAM-NATURE-CONN-HEX
in the invite x-header. For example, if the IE value is 00001010 in
the IAM, NSG will send X-FreeTDM-IAM-NATURE-CONN-HEX with value string
of "0A".
- Two characters are madatory. "0A" must be specified rather than "A". If
invalid format or characters are found, this x-header will be ommited.
- Valid characters include 0-9, A-F, and a-f
This feature allows ftmod_libpri to ignore calls with non-matching destination number.
You may want to use this on BRI PTMP lines (Point-to-MultiPoint),
to avoid conflicts between your FreeSWITCH server and other devices connected to the line.
The filter is disabled by default (all calls will be accepted),
setting one (or more) "local-number" parameters on the span configuration enables it.
Example configuration snippet:
<libpri_spans>
<span name="example01">
<!-- ... other span settings omitted ... -->
<param name="local-number" value="123456"/>
<param name="local-number" value="654321"/>
</span>
</libpri_spans>
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
readfile returns a value of type ssize_t (signed) and returns -1 if an
error occurs. In auth_readdb_internal, however, we were assigning the
return value of readfile to a variable of type size_t (unsigned), but
then testing this unsigned value to see if it was < 0, a
contradiction. We would thus simultaneously fail to report the error
in readfile and would end up with a corrupted length value.
sres_cached_answers_sockaddr is supposed to return ENOENT if no cached
records are found. Because of the missing return statement, however,
it would never do this and would instead return something very likely
to be garbage.
base64_d returns a value of size_t, which is unsigned. The value
therefore cannot be less than zero. The second check testing whether
it is >= INT_MAX is not a contradiction, but it doesn't make any sense
to check for this (as far as I can tell).
Converting these macros to functions declared static inline allow the
C type-checker to work and avoid warnings about unused expression
values. These warnings break the build with clang.
This avoid warnings about expressions with unused values. These
warnings break the build with clang.
An optimizing compiler should still inline these calls. If that turns
out not to happen on some platform, we could rename the functions used
internally and declare them static inline.
The implementation clears the context / state data from memory when it
is finished with it. Prior to this commit, however, it was actually
only clearing the first 4 bytes on x86 or 8 bytes on x86_64.
clang warns:
warning: argument to 'sizeof' in 'memset' call is the same
expression as the destination; did you mean to dereference it?
[-Wsizeof-pointer-memaccess]
Read the forward call indicator IE and print it into channel variable
ss7_iam_fwd_ind_hex. If this variable exists, put it in the x-header.
This implementation takes bits of A, CB, D, E, F, HG, I from the hex
value. Bits of KJ, L, P-M are not taken and set to 0.
The hex value is H-A-P-I, H is the highest bit to A, and next is P-I.
I is the lowest bit in the whole field, and H is the highest bit in
the whole field. Refer to Q.763 chapter 3.23.
In a sofia profile, you can now set the parameter tls-timeout to a
positive integer value which represents the maximum time in seconds
that OpenSSL will keep a TLS session (and its ephemeral keys) alive.
This value is passed to OpenSSL's SSL_CTX_set_timeout(3).
OpenSSL's default value is 300 seconds, but the relevant standard
(RFC 2246) suggests that much longer session lifetimes are
acceptable (it recommends values less than 24 hours).
Longer values can be useful for extending battery life on mobile
devices.
Signed-off-by: Travis Cross <tc@traviscross.com>
* fixed bug with remote hello-hash buffer being too small
* removed unused hello-hash storage in zrtp stream context
* fixed bug with libzrtp rendered "empty" SAS hash from incoming
SasRelay packet
* incremented libzrtp version number to 1.15
Signed-off-by: Travis Cross <tc@traviscross.com>
IE's heximal value.
- add <action application="export" data="freetdm_iam_fwd_ind_HEX=2301"/>
to dialplan with expected hex value. If the outgoing ftdm channel's
forward indicator needs to be changed, "export" needs to be put in the
incoming channel's dialplan to set this variable value accordingly.
- this implementation takes bits of A, CB, D, E, F, HG, I from the hex
value. Bits of KJ, L, P-M are not taken from the hex.
- How to calculate hex value with wanted bits:
. the hex value is H-A-P-I, H is the highest bit to A, and next is P-I.
I is the lowest bit in the whole field, and H is the highest bit in
the whole field. Refer to Q.763 chapter 3.23.
. use a text pad to fill in the bits with 0 and 1. eventually fill all
the 16 bits
. copy the binary value into a calculator and convert it to hex
It appears that libzrtp would prevent a non-passive client from
enrolling with a PBX because of a broken license mode comparison.
Thanks to Ivo Kutts for the report.
FS-4223 --resolve
native bridge mode
- This is supposed to be included in commit of
b324f86797. Somehow it's
not included in that commit. Without this change, the
REL receiving leg always stay in TERMINATING state when
received an incoming REL message.
Non-PRI_NEW_SET_API logging callbacks were only available in libpri-1.0 and older,
which also lacks PRI_IO_FUNCS (required) and wouldn't work anyway.
Explicitly check for both PRI_* feature defines at configure time and reject libpri
versions that lack them.
Remove the non-PRI_NEW_SET_API logging callbacks in ftmod_libpri.c.
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
Use the PRI_NEW_SET_API define provided by >=libpri-1.2 to distinguish
between old style and new style pri_set_error() and pri_set_message()
callback functions.
Improve message logging by using ftdm_log_chan() if per-span
data with a valid (d-)channel object is available.
NOTE: pri_get_userdata() returns NULL if pri is NULL.
This will reduce the horizontal space for libpri output a bit, but allows
us to see which span the message/error came from.
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
is down and recovered later
To re-produce this bug:
1. do CGB on one side
2. unplug signaling link cable
3. plug signaling link cable back
4. do CGU on the blocking side
5. cic state stay in RESTART for ever
Fix this problem by sending cic to SUSPENDED state after
receiving/sending CGU message
Timer-based b-channel tx gating won't work anyway, so remove all those
"#if 0"-ed bits of cruft.
Also remove the mISDN-specific timerfd_create() check in configure.
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
- When NSG receives INR from network, send back INF with calling
party category information IE and calling number information IE.
- Introduced a new global setting of "force-inr" for testing
purpose. Stinga generated INR/INF packets are not acceptable by
trillium stack since it misses call related information in the
packets. If configure force-inr to true in freetdm.conf.xml, when
NSG receives an incoming IAM, it'll send out INR packet regardless
of incoming IAM's IEs, and keep waiting for INF response from the
calling side.
- T.39 timer is introduced in order to handle INR timeout. The
default value of T.39 is 12 seconds and is configurable according
to spec.
- Only supports calling number IE and calling party category IE in
current fix. The customer only needs the calling number IE right now.
In ISUP spec, there are 6 optional IEs. NSG only supports calling
party number and calling category information IE since the other
IEs are not configurable in freetdm.conf.xml or included in IAM
message.
- In collect state, INR/INF implementation needs to work with existed
SAM messages. If NSG sent out INR and wait for SAM, collect state
check both INF received and enough dialed numbers received. If one
of these conditions are not met, it'll stay in collect state and wait
until either conditions met or timeout. After received INF and enough
dailed number, state moves to dailing and proceed as regular calls.
Re-bootstrapping is needed after this change. Previously CentOS 5
systems wouldn't build correctly without this file in tree, but we've
adjusted bootstrap to compensate.
resource-cleanup responsibilities clearly between the 2 channels involved in the bridge
- Each channel is responsible for clearning its own peer_data and event queue
at the end of the call (when moving to DOWN state)
- Each channel dequeues messages only from its own queue and enqueues messages
in the peer's queue, with the only exception being messages received before
the bridge is stablished (IAM for sure and possible SAM messages) because
if the bridge is not yet stablished the messages must be queued by the channel
in its own queue temporarily until the bridge is ready
- When the bridge is ready it is the responsibility of the incoming channel to
move the messages that stored temporarily in its own queue to the bridged peer queue
- During hangup, each channel is responsible for moving itself to DOWN. The procedure
however differs slightly depending on the hangup conditions
If the user requests hangup (ie, FreeSWITCH) the request will be noted by setting the
FTDM_CHANNEL_USER_HANGUP flag but will not be processed yet because call control is
driven only by the link messages (so no hangup from ESL or command line allowed)
When REL message comes, the channel receiving it must move to TERMINATING state and:
- If the user has not hangup yet (FTDM_CHANNEL_USER_HANGUP flag not set) then
notify the user via SIGEVENT_STOP and wait for the user to move to HANGUP
state by calling ftdm_channel_call_hangup() before sending RLC
- If the user did hangup already (FTDM_CHANNEL_USER_HANGUP flag is set) then
skip user notification and move to HANGUP state directly where the RLC message
will be sent
- On HANGUP state the RLC is sent and the channel is moved to DOWN, final state
The peer channel will forward the REL message and wait for RLC from the network, when
RLC is received the channel can move straight to DOWN itself because the peer channel
is completing its own shutdown procedure when it received the REL message
Thanks to Phil Zimmermann for the code and for the license exception
we needed to include it.
There remains some build system integration work to be done before
this code will build properly in the FreeSWITCH tree.
This fixes an issue where ss7 native bridge was accidentally enabled
any time two freetdm channels were bridged regardless of the freetdm_native_sigbridge
variable value.
Use the amount of audio data received in misdn_read() to determine how many
bytes we need to send to the b-channel (= how much free space is left
in the b-channel tx queue). (This is how libosmo-abis and LCR handle it too.)
A pipe is used as a poll()-able audio tx buffer (filled in misdn_write()):
FTDM_WRITE wait requests are currently poll()-ed on the input side of the pipe,
whereas FTDM_READ and _EVENT requests are poll()-ed on the b-channel socket itself.
For every N-bytes of audio data read from the b-channel in misdn_read(),
we try to get as much out of the tx pipe, convert it into the ISDN_P_B_RAW
format and send it to the b-channel socket.
If there's less than N-bytes left in the pipe, we fill the remaining buffer
with silence to avoid buffer underflows.
B-Channel handling overview:
- misdn_wait(FTDM_WRITE) on audio pipe
- misdn_write() put audio data into pipe
- misdn_wait(FTDM_READ) for next incoming mISDN
message on b-channel socket
- misdn_read() handle mISDN event, for PH_DATA_IND:
- Write data into channel buffer and convert
to a/u-law using misdn_convert_audio_bits()
- Try to fetch N-bytes from audio pipe
- If not enough bytes in pipe: fill remaining space with silence
- Convert audio to raw format
- Send to b-channel (PH_DATA_REQ)
Known problems / bugs / further investigation:
1. Bridge aborted by "Write Buffer 0 bytes Failed!" error from switch_core_io.c.
This is "fixed" by _not_ setting the b-channel sockfd to non-blocking mode.
2. Audio glitches (maybe caused by FTDM_WRITE misdn_wait() handling or blocking I/O on sockfd?)
3. misdn_read() EBUSY error messages from sending data to b-channel sockfd after enabling channel.
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
The former fixes a strange "bug" with hfcsusb, where a b-channel deactivation
on a inactive channel (caused by a reset cycle) would cause the port to
lock up and stop processing events.
NOTE: this still needs to be investigated further, but this workaround will
at least prevent it from breaking completely.
We'll now keep track of the channel activation state and not send any
PH_ACTIVATE_REQ / PH_DEACTIVATE_REQ requests, if the channel already has the
desired state.
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
Use POLLIN on the socket instead, the b-channel should be able
to write when there is something to read.
Several other projects handle it this way, e.g. libosmo-abis.
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
The code was improperly using peer_data as an indicator that the sigbridge
ss7 mode was enabled. The channel flag FTDM_CHANNEL_NATIVE_SIGBRDIGE should
be used instead
- The outgoing tdm leg should not move to UP until
after the IAM is sent at the end of the function
- The UP state should be processed immediately otherwise
the state processor is not run due to the way the main
ss7 processing loop currently works
Use FTDM_SIZE_FMT where needed, don't treat ftdm_event_t as an int
(even if the e_type enum is the first member), datalen vs. *datalen fix
and other warnings.
All reported by __check_printf() (GCC + __attribute__((format(printf,x,y))) ).
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
... that could cause segmentation faults.
Caught while working on __check_printf() support for ftdm_log().
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
For compilers that seem to do the wrong thing(tm).
Speculative fix for:
segfault at 1 ip b72145d3 sp b58f8bfc error 4 in libc-2.11.3.so
#0 0xb7a5d5d3 in vfprintf () from /lib/i686/cmov/libc.so.6
#1 0xb7a7cec7 in vasprintf () from /lib/i686/cmov/libc.so.6
#2 0xb7dd7c5b in switch_vasprintf (...)
#3 0xb6296de2 in ftdm_logger (...)
#4 0xb621625d in misdn_handle_mph_information_ind (...) at ftmod_misdn.c:658
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
Moises Silva
Jeff Lenk
Michael Jerris
Steve Underwood
Anthony Minessale
Matteo Brancaleoni
Ken Rice
Seven Du
Mathieu Rene
Stefan Knoblich
David Yat Sin
Robert Jongbloed
Travis Cross
Moises Silva
Brian West
Steve Underwood
Nenad Corbic
Kapil
SwK
root
James Zhang
Michael S Collins
Steve Underwood
Viktor Krykun
Gideon Sadan
Mitch Capper
Christopher Rienzo