|
|
@@ -0,0 +1,562 @@
|
|
|
+<!-- 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"/>
|
|
|
+ -->
|
|
|
+ <!--
|
|
|
+ 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>
|
|
|
+</configuration>
|
Mark Crane