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