71066559a8
Fix missing includes for GSM BS support
2013-03-11 17:05:24 +01:00
77d9102954
Add GSM full rate codec to LCR's source repository
...
There is no more need to download a seperate version of GSM full rate
(06.10) codec anymore.
2013-03-09 18:15:33 +01:00
0f0568583f
SIP: Extract IMSI from SIP URI
...
OpenBTS forwards IMSI via SIP name. In order to allow routing decision
by IMSI, the IMSI must be extracted from SIP name.
2013-01-06 08:58:16 +01:00
8719f21d8b
Fix: Correctly forward facility IE content
2013-01-06 06:33:56 +01:00
8a3a986487
Fix: Make action.cpp compile without mISDN/FXS support
2012-12-27 00:40:28 +01:00
2ea1d7ee45
Fix: Only screen caller ID 2 and redir ID when existing
...
Thanx to Wimpy for pointing to this bug.
2012-12-17 06:12:39 +01:00
b2dfcf34e6
Change Version to 1.14
2012-12-16 10:12:45 +01:00
4b85a2abcd
Added option to change DTMF decoding threshold level
...
If not given, the DSP modules' default value is used, rather than setting
it to 0. This was a bug.
2012-12-16 10:12:45 +01:00
e9b1625405
Fix: Disable DTMF dialing after first received KP (pulse) digit
...
Once a pulse digit is detected, it makes no sense to detect DTMF.
Pulses will create distortion with some phones, causing false
detection of DTMF tones.
2012-12-16 10:12:44 +01:00
acaf278f7f
Add FXS support
...
This requires FXS support to mISDN too.
2012-12-16 10:12:44 +01:00
fa1f601f49
Fix: 3PTY bridge must check, if other 3PTY member is mISDN or not
...
To make decision for mISDN bridge or lcr bridge, it it is required to
check both joins that share same 3PTY bridge.
2012-12-16 10:11:47 +01:00
84b78bea8c
chan_lcr: Replaced 'n' (no DTMF) option with 'D' (DTMF)
...
The option 'n' was actually broken. Now it is replaced, because
generated DTMF tones may cause delay to SIP connections.
2012-12-16 10:11:47 +01:00
705f2dac84
Fixed early audio with chan_lcr (Asterisk)
...
If progress message is received, go into proceeding state.
Send audio, if proceeding/alerting state, so RTP stream is sent in both
directions. This is essential when using NAT.
2012-12-16 10:11:47 +01:00
d2bcbfbaf0
Fixed version issue of chan_lcr
2012-12-16 10:11:47 +01:00
b4987df679
Updated MNCC interface
...
The structure of BEARER CAPABILITY has been expanded.
2012-12-16 10:11:47 +01:00
1fd1d5d17c
SIP: Allow early audio on incomming connections at SIP interface
...
In order to provide internal tones, a clock is used to generate
chunks of 160 samples. If no tones are provided and if audio is
bridged, it is forwarded as usual.
In order to provide early audio on SIP trunk, "tones yes" must be set
at interface.conf.
In order to receive early audio from SIP trunk, "earlyb yes" must be
set at interface.conf.
2012-12-16 10:11:47 +01:00
9e6a068f25
Fix: Set correct local RTP port
2012-12-16 10:11:47 +01:00
d3b4611440
Fix: Allow recording of audio for SIP/remote/GSM interfaces too
2012-12-16 10:11:47 +01:00
0562b894ff
Fix: Track notification messages at partyline too
...
This is required, so inactive parties will be marked as beeing "on hold".
These parties will be removed from the bridge, so the partyline is not
disturbed by hold music comming from inactive parties.
Thanx to Atul for pointing to this bug.
2012-12-16 10:11:47 +01:00
d19b20fa4e
Removed obsolete logging code
2012-12-16 10:11:46 +01:00
bdd274c5e8
Fix: Don't forward MESSAGE_TRAFFIC to endpoint instance
...
Thanx to Atul for catching this bug
2012-12-16 10:11:46 +01:00
0954d11aca
Fix: Append information (overlap dialing) to Asterisk's extension string
...
Signed-off-by: Andreas Eversberg <jolly@eversberg.eu>
2012-12-16 10:11:46 +01:00
b971fdeb62
Store caller/dialing info for calls via remote interfaces (chan_lcr)
2012-12-16 10:11:46 +01:00
cde9a763b1
Fix: Polling of file descriptors
...
It is only done when enabled by config or when any SIP interface is
created.
Thanx to Wimpy for catching this bug.
2012-12-16 10:11:46 +01:00
2bdb5135a3
Fix: chan_lcr must use right context attribute for Asterisk version >= 11
2012-12-16 10:11:46 +01:00
0d719518fb
Fix: Bind RTP/RTCP socket pairs correctly
2012-12-16 10:11:46 +01:00
a296b797ba
Fix: Always keep transmit timer on when mISDN channel is open
...
This way the buffer load is always calculated correctly.
2012-12-16 10:11:46 +01:00
29d2da58fc
Fix: Encoding of 3PTY result facility IE
2012-12-16 10:11:46 +01:00
c33007184d
Fix: Send tones/patterns/announcements for remote connections
2012-12-16 10:11:46 +01:00
68ccf0448d
Add screening of caller ID for remote (asterisk) connections
2012-12-16 10:11:46 +01:00
7f0d14c706
Cleanup: Make interface name be part of Port class
2012-12-16 10:11:46 +01:00
bd2aa91302
chan_lcr: Select remote interface by chan_lcr
...
Usage: Dial(LCR/<interface>/<digits>/<options>)
The interface must match the interface name in interface.conf. If omitted,
the first remote interface is used.
Example:
Dial(LCR/ast/123) will send a call to LCR and select remote interface
'ast'.
Dial(LCR//123) will send a call to LCR and select the first remote
interface.
Now it is possible to have multiple remote interfaces.
2012-12-16 10:10:34 +01:00
a28dde2f2b
Fix: Match complete string when filtering for interface
2012-12-16 10:10:34 +01:00
948b837669
Display source and destination interface at endpoint logging
2012-12-16 10:10:34 +01:00
19f537c691
Fix: Prevent Asterisk from aborting when delivering ast_frames
2012-12-16 10:10:34 +01:00
464ed1f284
Fix: LCR's DTMF detection will be enabled and used by default
...
Using 'n' option will disable it
Using 'a' option will disable it and use Asterisk's DTMF detection instead.
2012-12-16 10:10:34 +01:00
b5d6f1c72b
Fix: Asterisk DTMF detection works now
...
To enable, use option "a".
-> for calls from LCR use lcr_config(a) in extensions.conf
-> for calls to LCR use Dial(LCR/pbx/<number>/a)
2012-12-16 10:10:34 +01:00
680147f78c
Fix: chan_lcr will suppress audio traffic until ref is received
...
If no ref is received from LCR, the traffic may not be sent to LCR.
2012-12-16 10:10:34 +01:00
fcc787b14d
chan_lcr: Disabled bridge, because there is no concept right now.
2012-12-16 10:10:34 +01:00
72dada9cd5
Removed obsolete definition of media_type2name() from sip.h
2012-12-16 10:10:34 +01:00
49dd4be5b9
Fixed broken chan_lcr of last commit
2012-12-16 10:10:34 +01:00
4545bb054f
Fixed compiling of chan_lcr with Asterisk 1.6.2.2
2012-12-16 10:10:33 +01:00
7440c9a44d
Fix: Process tx-load when briding with jitter buffer disabled
2012-12-16 10:10:33 +01:00
71e76fd9e0
Updated default config examples
2012-12-16 10:10:33 +01:00
b6a3cd5a8d
Fixed parsing capability conditions
2012-12-16 10:10:33 +01:00
3f7ef909c9
Define prload of mISDN buffer by chan_lcr (required for fax)
...
Use q<ms> option to peload.
2012-12-16 10:10:33 +01:00
8b9bdad861
Bump version to 1.13
2012-12-16 10:10:33 +01:00
240562640f
Maintain states for remote socket connections
2012-12-16 10:10:33 +01:00
04fc928a2c
Implement 3PTY bridge of two 'join's.
2012-12-16 10:10:33 +01:00
30224b43e2
Add 3PTY facility to invoke conference call via functional protocol
2012-12-16 10:10:33 +01:00
Nick Vervelakis