<!-- Please refer to http://wiki.freeswitch.org/wiki/FreeTDM for further documentation -->
<!--
This is a sample FreeSWITCH XML configuration for FreeTDM
Remember you still need to configure freetdm.conf (no XML extension) in $prefix/conf/
directory of FreeSWITCH. The freetdm.conf (no XML extension) is a simple text file
definining the I/O interfaces (Sangoma, DAHDI etc). This file (freetdm.conf.xml) deals
with the signaling protocols that you can run on top of your I/O interfaces.
-->
<configuration name="freetdm.conf" description="FreeTDM Configuration">
<settings>
<param name="debug" value="0"/>
<!--<param name="hold-music" value="$${moh_uri}"/>-->
<!-- Analog global options (they apply to all spans)
Remember you can only choose between either call-swap
or 3-way, not both!
-->
<!--<param name="enable-analog-option" value="call-swap"/>-->
<!--<param name="enable-analog-option" value="3-way"/>-->
<!--
Refuse to load the module if there is configuration errors
Defaults to 'no'
-->
<!--<param name="fail-on-error" value="no"/>-->
</settings>
<!-- Sample analog configuration (The analog_spans tag is for ftmod_analog) -->
<analog_spans>
<!-- The span name must match the name in your freetdm.conf -->
<span name="myAnalog">
<!--<param name="hold-music" value="$${moh_uri}"/>-->
<!--
3-way allows you to flash your FXS line and dial
another number and put all the parties in a conference
call-swap allows you to flash your FXS line and swap
between one call and another
Remember you can only choose between either call-swap
or 3-way, not both!
<param name="enable-analog-option" value="call-swap"/>
<param name="enable-analog-option" value="3-way"/>
-->
<!-- Tones are defined in tones.conf
This setting is very important for analog lines to
work properly
-->
<param name="tonegroup" value="us"/>
<!-- How much time to wait for digits (in FXS lines) -->
<param name="digit-timeout" value="2000"/>
<!-- Maximum number of digits to wait for (in FXS lines) -->
<param name="max-digits" value="11"/>
<!-- whether you want to wait for caller id -->
<param name="enable-callerid" value="true"/>
<!-- How much to wait for dial tone (0 if you just want to dial out immediately without waiting for dial tone) -->
<!--<param name="wait-dialtone-timeout" value="5000"/>-->
<!-- whether you want to enable callwaiting feature -->
<!--<param name="callwaiting" value="true"/>-->
<!-- whether you want to answer/hangup on polarity reverse for outgoing calls in FXO devices
and send polarity reverse on answer/hangup for incoming calls in FXS devices -->
<!--<param name="answer-polarity-reverse" value="false"/>-->
<!--<param name="hangup-polarity-reverse" value="false"/>-->
<!--
Minimum delay (in milliseconds) required between an answer polarity reverse
and hangup polarity reverse in order to assume the second polarity reverse is a real hangup
<param name="polarity-delay" value="600"/>
-->
<!-- Retrieve caller id on polarity reverse -->
<!--
<param name="polarity-callerid" value="true"/>
-->
<!-- regex to stop dialing when it matches -->
<!--<param name="dial-regex" value="5555"/>-->
<!-- regex to stop dialing when it does not match -->
<!--<param name="fail-dial-regex" value="^5"/>-->
<!-- FreeSWITCH dialplan type and context to send the calls -->
<param name="dialplan" value="XML"/>
<param name="context" value="default"/>
</span>
</analog_spans>
<!--
openr2 (MFC-R2 signaling) spans (ftmod_r2)
In order to use this type of spans your FreeTDM must have been compiled with ftmod_r2 module.
The module is compiled if the openr2 library is present when running the ./configure script
in the FreeTDM source code
MFC-R2 signaling has lots of variants from country to country and even sometimes
minor variants inside the same country. The only mandatory parameters here are:
variant, but typically you also want to set max_ani and max_dnis.
IT IS RECOMMENDED that you leave the default values (leaving them commented) for the
other parameters unless you have problems or you have been instructed to change some
parameter. OpenR2 library uses the 'variant' parameter to try to determine the
best defaults for your country. If you want to contribute your configs for a particular
country send them to the e-mail of the primary OpenR2 developer that you can find in the
AUTHORS file of the OpenR2 package, they will be added to the samples directory of openr2.
-->
<r2_spans>
<span name="wp1" cfgprofile="testr2">
<!--
MFC/R2 variant. This depends on the OpenR2 supported variants
A list of values can be found by executing the openr2 command r2test -l
some valid values are:
mx (Mexico)
ar (Argentina)
br (Brazil)
ph (Philippines)
itu (per ITU spec)
-->
<param name="variant" value="mx"/>
<!-- switch parameters (required), where to send calls to -->
<param name="dialplan" value="XML"/>
<param name="context" value="default"/>
<!--
Max amount of ANI (caller id digits) to ask for
<param name="max_ani" value="4"/>
-->
<!--
Max amount of DNIS to ask for
<param name="max_dnis" value="4"/>
-->
<!-- Do not set parameters below this line unless you desire to tweak it because is not working -->
<!--
Whether or not to get the ANI before getting DNIS (only affects incoming calls)
Some telcos require ANI first some others do not care, if default go wrong on
incoming calls, change this value
<param name="get_ani_first" value="yes"/>
-->
<!--
Caller Category to send. Accepted values:
- national_subscriber
- national_priority_subscriber
- international_subscriber
- international_priority_subscriber
- collect_call
Usually national_subscriber (the default) works just fine
<param name="category" value="national_subscriber"/>
-->
<!--
Brazil uses a special calling party category for collect calls (llamadas por cobrar)
instead of using the operator (as in Mexico). The R2 spec in Brazil says a special GB tone
should be used to reject collect calls. If you want to ALLOW collect calls specify 'yes',
if you want to BLOCK collect calls then say 'no'. Default is to block collect calls.
(see also 'double_answer')
<param name="allow_collect_calls" value="yes"/>
-->
<!--
This feature is related but independent of allow_collect_calls
Some PBX's require a double-answer process to block collect calls, if
you ever have problems blocking collect calls using Group B signals (allow_collect_calls=no)
then you may want to try with double_answer=yes, this will cause that every answer signal
is changed to perform 'answer -> clear back -> answer' (sort of a flash)
(see also 'allow_collect_calls')
<param name="double_answer" value="yes"/>
-->
<!--
This feature allows to skip the use of Group B/II signals and go directly
to the accepted state for incoming calls
<param name="immediate_accept" value="yes"/>
-->
<!--
Skip request of calling party category and ANI
<param name="skip_category" value="yes"/>
-->
<!--
Brazil use a special signal to force the release of the line (hangup) from the
backward perspective. When forced_release=no, the normal clear back signal
will be sent on hangup, which is OK for all mfcr2 variants I know of, except for
Brazilian variant, where the central will leave the line up for several seconds (30, 60)
which sometimes is not what people really want. When forced_release=yes, a different
signal will be sent to hangup the call indicating that the line should be released immediately
<param name="forced_release" value="yes"/>
-->
<!--
Whether or not report to the other end 'accept call with charge'
This setting has no effect with most telecos, usually is safe
leave the default (yes), but once in a while when interconnecting with
old PBXs this may be useful.
Concretely this affects the Group B signal used to accept calls
<param name="charge_calls" value="yes"/>
-->
<!--
MFC/R2 value in milliseconds for the MF timeout. Any negative value
means 'default', smaller values than 500ms are not recommended
and can cause malfunctioning. If you experience protocol error
due to MF timeout try incrementing this value in 500ms steps
<param name="mfback_timeout" value="1500"/>
-->
<!--
MFC/R2 value in milliseconds for the metering pulse timeout.
Metering pulses are sent by some telcos for some R2 variants
during a call presumably for billing purposes to indicate costs,
however this pulses use the same signal that is used to indicate
call hangup, therefore a timeout is sometimes required to distinguish
between a *real* hangup and a billing pulse that should not
last more than 500ms, If you experience call drops after some
minutes of being stablished try setting a value of some ms here,
values greater than 500ms are not recommended.
BE AWARE that choosing the proper protocol variant parameter
implicitly sets a good recommended value for this timer, use this
parameter only when you *really* want to override the default, otherwise
just comment out this value.
<param name="metering_pulse_timeout" value="1000"/>
-->
<!--
WARNING: advanced users only! I really mean it
this parameter is commented by default because
YOU DON'T NEED IT UNLESS YOU REALLY GROK MFC/R2
READ COMMENTS on doc/r2proto.conf in openr2 package
for more info
<param name="advanced_protocol_file" value="/usr/local/freeswitch/conf/r2proto.conf"/>
-->
<!-- USE THIS FOR DEBUGGING MFC-R2 PROTOCOL -->
<!--
Where to dump advanced call file protocol logs
<param name="logdir" value="$${base_dir}/log/mfcr2"/>
-->
<!--
MFC/R2 valid logging values are: all,error,warning,debug,notice,cas,mf,nothing
error,warning,debug and notice are self-descriptive
'cas' is for logging ABCD CAS tx and rx
'mf' is for logging of the Multi Frequency tones
You can mix up values, like: loglevel=error,debug,mf to log just error, debug and
multi frequency messages
'all' is a special value to log all the activity
'nothing' is a clean-up value, in case you want to not log any activity for
a channel or group of channels
BE AWARE that the level of output logged will ALSO depend on
the value you have in FreeSWITCH logging configurations, if you disable output FreeSWITCH
then it does not matter if you specify 'all' here, nothing will be logged
so FreeSWITCH has the last word on what is going to be logged
<param name="logging" value="debug,notice,warning,error,mf,cas"/>
-->
<!--
whether or not to drop protocol call files into 'logdir'
<param name="call_files" value="yes"/>
-->
<!--
Use only for very technical debugging
This is the size (if 0, dumps are disabled) of MF dump files. MF dump files
are audio files that are dumped when a protocol error occurs.
The files are dumped in whatever you set in the logdir parameter.
Value -1 uses a default recommended size (which stores 5 seconds of audio)
<param name="mf_dump_size" value="-1"/>
-->
</span>
</r2_spans>
<!-- Sangoma ISDN PRI/BRI spans. Requires libsng_isdn to be installed -->
<sangoma_pri_spans>
<span name="wp1">
<!--
Switch emulation/Variant
Possible values are:
national
4ess
5ess
qsig
euroisdn
ntt
<param name="switchtype" value="national"/>
-->
<!--
Signalling
Possible values are:
net
cpe
<param name="signalling" value="cpe"/>
-->
<!--
Overlap - whether to support overlap receive
Possible values are: Yes/No
<param name="overlap" value="yes"/>
-->
<!--
Facility - whether to support facility messages
Possible values are: Yes/No
<param name="facility" value="yes"/>
-->
<!--
Minimum Digits
In overlap receive mode.
Minimum number of digits to receive before sending notification
to the dialplan
Possible values are: <Any digit>
<param name="min-digits" value="8"/>
-->
<!--
TEI - default value for Terminal Equipment Identifier.
Used in Point-to-point connections
Possible values are: <1-127>
<param name="tei" value="0"/>
-->
<!--
Type of Number (TON)
Set the TON on outbound calls
Possible values are:
unknown
international
national
network-specific
subscriber-number
abbreviated-number
<param name="outbound-called-ton" value="unknown"/>
<param name="outbound-calling-ton" value="unknown"/>
<param name="outbound-rdnis-ton" value="unknown"/>
-->
<!--
Numbering Plan Indendification (NPI)
Set the NPI on outbound calls
Possible values are:
unknown
isdn
data
telex
national
private
reserved
<param name="outbound-called-npi" value="unknown"/>
<param name="outbound-calling-npi" value="unknown"/>
<param name="outbound-rdnis-npi" value="unknown"/>
-->
<!--
Bearer Capability - Transfer Capability
Set the Bearer Capability - Transfer Capability on outbound calls
Possible values are:
speech
unrestricted-digital-information
restricted-digital-information
3.1-Khz-audio
7-Khz-audio
15-Khz-audio
video
<param name="outbound-bc-transfer-cap" value="speech"/>
-->
<!--
Bearer Capability - User Layer 1
Set the Bearer Capability - User Layer 1 on outbound calls
Possible values are:
V.110
ulaw
alaw
<param name="outbound-bc-user-layer1" value="speech"/>
-->
<!--
Channel Restart Timeout
If we do not receive a RESTART message within this timeout on link
UP, we will send a channel restart.
<param name="channel-restart-timeout" value="20"/>
-->
<!--
Local Number (MSN)
On incoming calls, we will only respond to this call if
the Called Party Number matches this value.
Note: Up to 8 local numbers can be added per span.
<param name="local-number" value="9054741990"/>
-->
<!--
Facility Timeout
Amount of time to wait for the FACILITY message after a SETUP message is
received
<param name="facility-timeout" value="1"/>
-->
<!--
Transfer Timeout
Amount of time to wait for the remote switch to respond to a transfer request
<param name="transfer-timeout" value="20"/>
-->
<!--
AT&T Transfer - Remove DTMF
Whether to remove DTMF tones received from remote switch when performing
AT&T Transfer.
<param name="att-remove-dtmf" value="yes/no"/>
-->
<!--
Facility Information Element Decoding
Whether to decode contents within Facility IE. You should only disable this option if your custom application has its own Facility IE decoding.
<param name="facility-ie-decode" value="yes/no"/>
-->
<!--
Ignore cause value
When using 5ESS switchtype, whether or not do initiate disconnects based on cause code.
<param name="ignore-cause-value" value="yes/no"/>
-->
<!--
Trace (Interpreted)
Whether or not to enable Q921/Q931 trace on start
<param name="q931-trace" value="yes/no"/>
<param name="q921-trace" value="yes/no"/>
-->
<!--
Trace (Raw)
Whether or not to enable Q921/Q931 trace on start
<param name="q931-raw-trace" value="yes/no"/>
<param name="q921-raw-trace" value="yes/no"/>
-->
<!--
Force sending complete
Will add Sending Complete IE to outgoing SETUP message
By default, enabled on EuroISDN, disabled on US variants.
<param name="force-sending-complete" value="yes/no"/>
-->
<!--
Early Media Override
Assume early media is available, even if Q.931 message does not have
progress indicator IE = in-band data ready
Possible values
on-proceed
on-progress
on-alert
<param name="early-media-override" value="on-alert"/>
-->
<!--
Invert Channel ID Invert Bit
Invert the Channel ID Extend Bit
<param name="chan-id-invert-extend-bit" value="yes/no"/>
-->
<!--
CID Name transmit method
How to transmit Caller ID Name
Possible values:
display-ie
user-user-ie
facility-ie
default (will transmit CID-Name based on variant)
<param name="cid-name-transmit-method" value="default"/>
-->
<!--
CID Name transmit
Whether to transmit Caller ID Name
Possible values:
yes - always send CID-name
no - nether send CID-name
default (will transmit CID-Name based on variant)
<param name="cid-name-transmit-method" value="default"/>
-->
<!--
Send CONNECT ACK
Whether to send a Connect Ack
Possible values:
yes - send connect ack
no - do not send connect ack
default (will transmit Connect Ack based on variant + signaling)
<param name="send-connect-ack" value="yes"/>
-->
<!--
Q.931 Timers in seconds
Override default Q.931 values
timers:
timer-t301
timer-t302
timer-t303
timer-t304
timer-t305
timer-t306
timer-t307
timer-t308
timer-t310
timer-t312
timer-t313
timer-t314
timer-t316
timer-t318
timer-t319
timer-t322
<param name="timer-t301" value="10"/>
-->
</span>
</sangoma_pri_spans>
<!--
PRI passive tapping spans. Requires patched version from libpri at http://svn.digium.com/svn/libpri/team/moy/tap-1.4
You must also configure FreeTDM with "-with-pritap" (see ./configure help for details)
-->
<pritap_spans>
<span name="tapped1">
<!-- The peer span name used to tap the link -->
<param name="peerspan" value="tapped2"/>
<!--
Whether to mix the audio from the peerspan with the audio from this span
This is most likely what you want (and therefore the default) so you can hear
the full conversation being tapped instead of just one side
-->
<!-- <param name="mixaudio" value="yes"/> -->
<!-- switch parameters (required), where to send calls to -->
<param name="dialplan" value="XML"/>
<param name="context" value="default"/>
</span>
<span name="tapped2">
<!-- This span is linked to "tapped1" through its peerspan parameter -->
<param name="peerspan" value="tapped1"/>
<!-- <param name="mixaudio" value="yes"/> -->
<!-- switch parameters (required), where to send calls to -->
<param name="dialplan" value="XML"/>
<param name="context" value="default"/>
</span>
</pritap_spans>
<!--
GSM spans (libwat must be installed when configuring the freetdm build)
-->
<gsm_spans>
<span name="gsm01">
<!-- where to send inbound calls to -->
<param name="dialplan" value="XML" />
<param name="context" value="module1" />
<!--
GSM module type
Accepted values: "telit", "telit-gc864", "telit-he910", "telit-cc864", "telit-de910", "motorola"
-->
<param name="moduletype" value="telit-gc864" />
<!--
Debug mask (accepted values: all, uart_raw, uart_raw, call_state, span_state, at_parse, at_handle, sms_encode, sms_decode, none)
comma-separated values are accepted to combine those levels
-->
<param name="debug" value="all" />
<!--
Whether to enable HW DTMF in the hardware module
Accepted values: true, generate, detect, false
-->
<param name="hwdtmf" value="true" />
<!--
This enables conditional forwarding on module startup
-->
<!-- <param name="conditional-forwarding-prefix" value="*71" /> -->
<!-- <param name="conditional-forwarding-number" value="123456789" /> -->
<!--
This enables immediate forwarding logic to simulate hunt groups
The syntax for immediate-forwarding-numbers ia a comma-separated
list of elements in the form [<span-name>:]<number>
if span-name is specified, the span will be checked for availability
before enabling forwarding to that span number
The span-name must be a defined freetdm span name
If the span-name is not specified then only first number specified is used for
forwarding whenever this span is busy
-->
<!-- <param name="immediate-forwarding-prefix" value="*72" /> -->
<!-- <param name="immediate-forwarding-numbers" value="gsm02:123456789" /> -->
<!-- Number to dial to disable forwarding when the call ends (if immediate forwarding was enabled) -->
<!-- <param name="disable-forwarding-number" value="*73" /> -->
<!-- Startup raw AT commands to run, you can repeat this parameter as many times as needed -->
<!-- <param name="startup-command" value="AT$GPSP=1" /> -->
<!-- <param name="startup-command" value="AT$GPSAT=1" /> -->
</span>
</gsm_spans>
</configuration>
