. 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
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
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.
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
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.
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
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>
David Yat Sin