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Configure your Sofia gateway for IPComms SIP trunking
Create a new file: /etc/freeswitch/sip_profiles/external/ipcomms.xml
<include>
<gateway name="ipcomms">
<param name="username" value="YOUR_USERNAME"/>
<param name="password" value="YOUR_PASSWORD"/>
<param name="realm" value="sip.ipcomms.net"/>
<param name="proxy" value="sip.ipcomms.net"/>
<param name="register" value="true"/>
<param name="register-transport" value="udp"/>
<param name="expire-seconds" value="300"/>
<param name="caller-id-in-from" value="false"/>
</gateway>
</include>Add to your dialplan (e.g., default.xml):
<extension name="outbound-ipcomms">
<condition field="destination_number" expression="^1?(\d{10})$">
<action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>
<action application="bridge" data="sofia/gateway/ipcomms/1$1"/>
</condition>
</extension>Route incoming calls from your DID:
<extension name="inbound-did">
<condition field="destination_number" expression="^(\d{10})$">
<action application="set" data="domain_name=$${domain}"/>
<action application="transfer" data="1000 XML default"/>
</condition>
</extension>Apply changes via fs_cli:
fs_cli -x "sofia profile external rescan"
fs_cli -x "sofia status gateway ipcomms"Gateway should show REGED status. Make a test call!
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We recommend G.711 (PCMU/PCMA) for best quality. G.729 is also supported if you have licenses.
Yes! Configure your gateway with register-transport="tls" and use SRTP for encrypted media.
Absolutely. Create multiple gateway definitions and use the failover bridge syntax in your dialplan.
Yes! T.38 is fully supported. Configure your spandsp module as usual.
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