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SIP Trunking for FreeSWITCH

High-performance SIP trunks built for FreeSWITCH deployments. Sofia gateway configurations, XML dialplan examples, and support from engineers who speak FreeSWITCH.

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High Performance

Built for high-concurrency deployments

XML Examples

Ready-to-use dialplan and gateway configs

TLS + SRTP

Full encryption for signaling and media

Expert Support

Engineers who understand mod_sofia

FreeSWITCH Setup Guide

Configure your Sofia gateway for IPComms SIP trunking

Create Gateway Configuration

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="yourdomain.s1.ipcomms.net"/>
    <param name="proxy" value="yourdomain.s1.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>

Outbound Dialplan

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>

Inbound Dialplan

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>

Reload Sofia

Apply changes via fs_cli:

fs_cli -x "sofia profile external rescan"
fs_cli -x "sofia status gateway ipcomms"

Verify Registration

Gateway should show REGED status. Make a test call!

Simple Pricing

No hidden fees. Scale as you need.

Metered

Predictable costs for steady volume

ChannelNo fee Outbound$0.010/min InboundFree

Pay-As-You-Go

No monthly fees, just usage

Monthly fee$0 Outbound$0.010/min InboundFree
View full pricing

FreeSWITCH FAQ

What codec should I use?

We recommend G.711 (PCMU/PCMA) for best quality. G.729 is also supported if you have licenses.

Do you support TLS/SRTP?

Yes! Configure your gateway with register-transport="tls" and use SRTP for encrypted media.

Can I use multiple gateways for failover?

Absolutely. Create multiple gateway definitions and use the failover bridge syntax in your dialplan.

Do you support T.38 fax?

Yes! T.38 is fully supported. Configure your spandsp module as usual.

Next step

$0.010/min outbound to the US and Canada, $0.009/min inbound, $0 per channel, no contracts.

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