Make it possible to configure the transmit power. These settings
will be in effect immediately (there is no lock between the two
threads but it is a read only).
When an endpoint is allocated we do not have enough resources to
connect it again and this situation is permanent as long as the
endpoint is connected.
Returning 502 might help some switches to release stale audio resources.
Playing tones should be printable ascii characters, begin by checking
for NULL (as it happened before) as the first illegal tone. This is
an attempt to make the API more robust and detect errors more early.
Because of the issue parsing the MGCP request a '\0' was added to
the queue. This lead to the dtmf scheduler believing that a play
was in progress while the playing code didn't play anything. This
lead to the queue to be stuck and nothing being played at all.
Return the number of tones that should be played and stop using
strlen on the tones.
The RQNT message might not contain a 'S:' line with the actual tone
to play. Instead of calling the callback with the 0 as tone just leave
early.
Example:
X: 6B9519B88F0
R: D/[0-9#*](N), G/ft, fxr/t38
Merged from the OpenBSC development tree
For some equipment it is the easiest to patch the assignment
complete message transported to the MSC. Add a VTY config to
enable this patching, create a testcase that tests that the
original message is truncated. The setting of the VTY option
has been manually tested. The entire system has not been end
to end tested.
Conflicts:
src/vty_interface.c
* Include stdlib.h before the snmp headers to have a free declaration
* Use sprintf(dest, "%s", str) to avoid format string attacks
* Avoid bogus assignment. This pattern was used for marking something
as unused in the past.
This is a hot fix to make CIC reading (and later status) work on
big endian machines. There might be a more elegant way to do it
and I will explore this later.
This way one can check the logs if this code path is hit. The
logging normally only occurs for the two messages we are interested
in. It doesn't make sense to reset with a range of 0 so the code
has been re-ordered.
GCC 3.x on PowerPC correctly highlights that the code is fishy.
Re-reading the RFC 3550 shows that we should subtract it and then
we are in the 16bit range. The probation and re-sync code is still
missing.
GCC:
mgcp/mgcp_network.c:200: warning: comparison is always true due to limited range of data type
For the ISUP/MGCP handling we will need the same code, extract it
from the msc_connection. For the reading code callback is introduced
that will pass the MGCP message to the higher layer.
The RSIP has morphed from a global reset, to a per trunk reset and now
it is possible to reset specific ranges on a trunk. This will be used by
the ISUP filter code in the STP.
For legacy range == -1 will be used. This will reset all endpoints
on the trunk. Use OSMO_MAX on endpoint and number of endpoints in case
number_endpoints is 0.
This code will now free everything from the endpoint to endpoint + range
including endpoint+range.
In case the endpoint is allocated and the VTY is used to force it
to be freed use the realloc_cb to make sure that any resources
allocated to it will be freed. The mgcp_ss7 binary will then call
mgcp_ss7_endp_free to free audio encoder resources.
There were several changes in the upstream code. These include
statistics, DTMF/RQNT, changes in the parsing code and re-transmission
handling. The last item is the main reason to do the merge now.
Create a simple queue for pending DTMF tones, play them using the
MTN API, and then send the next tones once the playback is complete.
The callback and scheduling is done from the same context so no locking
needs to be done.
Introduce a callback for the request and forward the signalrequest
to the callback. This is not a full implementation of MGCP RQNT.
Manual merge and backport from OpenBSC.
In case MtnSaAllocate failed the code did not restore the audio routing
when destroying the endpoint. Move the hw_maybe_loop_endp out of the if
statement.
This got broken in the previous commit. Use the hw_dsp_port we assigned
to the endpoint instead of the dsp_resource that is one ahead. For the
non loop-on-idle configuration audio routing appears to be broken.
* Change the assigning dsp_resource to start with one and increase
after usage and not before usage (so it becomes the next free
port).
* Subtract one from dsp_port after we have selected the path. In the
longer run I should make dsp_resources zero based all the way.
For a linkset define where SCCP/ISUP should be send. This config
should probably move up to the application part when real work on
the routing is done. Right now the sccp_opc/sccp_dpc need to stay
inside the mtp_layer3.c to be able to send a TFA for the reachable
OPC and it is easier to keep both (dpc/opc) in the same file.
Holger Hans Peter Freyther