Utilisant Asterisk comme une passerelle du fournisseur

Asterisk ⟷ FreeSWITCH

Bien sûr, vos besoins peuvent être différents, mais c'est un bon départ. - J'ai une paire de serveurs avec une connexion privée, vous devrez peut-être adapter des mesures d'authentification, mais cela devrait illustrer les bases de l'interaction en arrière et en avant, la transition vers le bon contexte, etc. .

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pjsip.conf

[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)

extension.conf

[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)

FreeSWITCH

conf / autoload_configs / acl.conf.xml

Ajoutez-le à l'intérieur

            <network-lists>

          

:

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

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

</list>

conf / sip_profiles / asterisk.xml

<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>

dialplan.xml

Adapter et ajouter à DIALPLAN

:

<extension name="to-asterisk">

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

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

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

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

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

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

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

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

</condition>

</extension>

</network-lists>