Używanie Asterisk jako bramy do Twojego usługodawcy internetowego

[tel]

type=transport

protocol=udp

bind=10.8.0.2 # set asterisk's IP -- bind to this address[acl]

type=acl

deny=0.0.0.0/0

permit=10.8.0.3/32 # allow only calls from freeswitch who is on 10.8.0.3 see above deny[fs]

type=identify

endpoint=fs

match=10.8.0.3 # identify/auth traffic from freeswitch by its IP[fs]

type=endpoint # set options for endpoint we identified just above

trust_id_inbound=yes

trust_id_outbound=yes

aors=fs

context=from-internal ## WHERE DO CALLS FROM FREESWITCH TO ASTERISK GO?

allow=!all,g722,ulaw

transport=tel[fs]

type=aor

contact=sip:10.8.0.3 # address-of-record to find freeswitch so can dial to fs without it registering with us (this is fed up to [fs] type=endpoint via its aors above so calls to Dial(PJSIP/1234@fs) dials 1234 on FreeSWITCH 10.8.0.3)

[from-internal]

include = toFreeSWITCH[toFreeSWITCH]

exten = _N11!,1,Dial(PJSIP/${EXTEN}@fs)

exten = _0!,1,Dial(PJSIP/${EXTEN}@fs)

exten = _3XX,1,Dial(PJSIP/${EXTEN}@fs)

exten = _1NXXNXXXXXX,1,Dial(PJSIP/${EXTEN}@fs)

exten = _508NXXXXXX,1,Dial(PJSIP/${EXTEN}@fs)

exten = _774NXXXXXX,1,Dial(PJSIP/${EXTEN}@fs)

           <network-lists>

         

<list name="asterisk" default="deny">

    <node type="allow" cidr="10.8.0.2/32"/>

</list>

<profile name="asterisk">

        <gateways>

            <gateway name="asterisk">

                <param name="username" value="freeswitch"/>

                <param name="realm" value="your-asterisk-domain"/>

                <param name="password" value="unused-but-required-field"/>

                <param name="from-domain" value="your-asterisk-domain"/>

                <param name="proxy" value="10.8.0.2"/><!-- ASTERISK ADDRESS -->

                <param name="register" value="false"/><!-- INSTEAD WE SET AOR IN PJSIP.CONF -->

                <param name="cid-type" value="pid"/>

                <param name="rfc-5626" value="true"/>

              </gateway>

        </gateways>

        <settings>

                <param name="apply-inbound-acl" value="asterisk"/>

                <param name="auth-calls" value="false"/>

                <param name="context" value="public"/><!-- WHERE DO CALLS FROM ASTERISK COME INTO? -->

                <param name="rtp-ip" value="10.8.0.3"/><!-- this FreeSWITCH MEDIA IP -->

                <param name="sip-ip" value="10.8.0.3"/><!-- this FreeSWITCH SIP IP -->

        </settings>

</profile>

<extension name="to-asterisk">

    <condition field="destination_number" expression="^([2-9]11|1?[2-9]\d{2}[2-9]\d{6}|3\d{2})$">

        <action application="set" data="dialed_extension=$1"/>

        <action application="export" data="dialed_extension=$1"/>

        <action application="set" data="call_timeout=30"/>

        <action application="set" data="hangup_after_bridge=true"/>

        <action application="set" data="continue_on_fail=true"/>

        <action application="export" data="rtp_secure_media=false"/>

        <action application="bridge" data="sofia/gateway/asterisk/${destination_number}"/>

    </condition>

</extension>