Documentation

Connect FreeSWITCH to Open Voice Shield

Add a sofia gateway for sip.ovs.telecomsxchange.com with register false, whitelist your server's public IP on the Network page, and bridge the calls you want analyzed to sofia/gateway/ovs: no credentials, G.711 in the INVITE. The verdict appears after hang-up.

Before you start#

• An Open Voice Shield account. Signup is self-serve and starts you with US$5 of credit.
• The public IP address your FreeSWITCH sends SIP from, the address after NAT, not the one on its interface.
• A destination, the carrier or SBC the calls were already going to. We relay each accepted call on to it.
• Firewall: UDP or TCP 5060 to the ingress, and UDP 30000–40000 open in both directions for RTP. See the firewall rules.

Step 1: whitelist your server's IP#

Open the Network page in the portal and add the public address your FreeSWITCH sends from as a CIDR entry, a single host is a /32. The whitelist is the authentication: there are no credentials, and an INVITE from any other address is answered 403.

Step 2: add the gateway#

The connection is a standard sofia gateway on your external profile with registration switched off, the same shape as any IP-authenticated carrier you already configure.

<include>
  <gateway name="ovs">
    <param name="proxy" value="sip.ovs.telecomsxchange.com:5060"/>
    <param name="register" value="false"/>
  </gateway>
</include>

Verified against the FreeSWITCH 1.10 sofia gateway documentation. Adapt the profile and file layout to your build, then run sofia profile external rescan.

• register=false matters: nothing registers in either direction, and a REGISTER sent to the ingress is answered 403.
• Offer G.711 first on this leg. absolute_codec_string=PCMU,PCMA in the bridge below pins it. Anything else is relayed but produces no transcript and no verdict.
• Your FreeSWITCH offers SDP in the initial INVITE by default. Keep it that way. An INVITE without SDP is answered 488.

Step 3: bridge every call through the gateway#

Bridge your outbound pattern to the new gateway. The dialled number and caller ID pass through unchanged, and the call leaves us to your configured destination.

<extension name="all-via-ovs">
  <condition field="destination_number" expression="^(\d+)$">
    <action application="bridge" data="{absolute_codec_string=PCMU,PCMA}sofia/gateway/ovs/$1"/>
  </condition>
</extension>

Or bridge one call in ten#

You do not have to move all your traffic, sampling is a supported pattern. A mod_distributor list picks a gateway by weight: one node in ten is the analyzed hop, the other nine are your direct carrier gateway.

<configuration name="distributor.conf" description="Weighted sampling">
  <lists>
    <list name="ovs_sample" total-weight="10">
      <node name="ovs" weight="1"/>
      <node name="carrier" weight="9"/>
    </list>
  </lists>
</configuration>

<extension name="sampled-via-ovs">
  <condition field="destination_number" expression="^(\d+)$">
    <action application="set" data="ovs_gw=${distributor(ovs_sample)}"/>
    <action application="bridge" data="{absolute_codec_string=PCMU,PCMA}sofia/gateway/${ovs_gw}/$1"/>
  </condition>
</extension>

Verified against the FreeSWITCH 1.10 mod_distributor documentation. Configure the node names to match your gateway names, and enable the module in modules.conf.xml.

Step 4: place a test call#

Dial a real number over the new route and talk on both legs for at least 30 seconds. A leg that never speaks produces a one-sided transcript. After hang-up the call appears in Calls in the console with its recording, per-channel transcript and verdict, usually within a minute.

• Rejected 403? The source address is not whitelisted. Compare your own SIP trace with the CIDR entries on the Network page.
• One-way audio? A firewall rule on UDP 30000–40000, not a signalling problem.
• A recording but no verdict? The call negotiated a codec we do not decode (G.711, G.722, G.729 and G.723.1 are supported): pin absolute_codec_string on the bridge.

Anything else: Troubleshooting lists every rejection code and the check that clears it.

Frequently asked questions#

Ready to try it? Create an account. Signup is self-serve and comes with US$5 of credit, enough to analyze your first test calls.