Installation instructions for FreeSWITCH

DID Logic regional gateways:

Regional gateway	FQDN	IP Address
Primary regional gateway	sip.se.didlogic.net	5.150.254.205
Secondary regional gateway	sip.nl.didlogic.net	185.118.63.77

Note: Carefully select a primary gateway with minimal ping and a failover option.

Configuring SIP Trunk on the DID Logic portal

Creating SIP Account

On the "SIP" tab click "Create account", then

1. Configure Device Name (optional). Device names are especially useful when you manage multiple SIP accounts.
2. Set password. Remember that strong passwords are a critical element of account security.
3. Repeat password.
4. Set Caller ID associated with SIP account. Only DIDs purchased through DID Logic will be available on the dropdown list.
5. ‘Send inbound didinfo’ option should be enabled when registering a PBX or an SBC. This feature instructs DID Logic gateways to replace the SIP Account number in the SIP URI and ‘To:’ headers with the originally dialed number (DID). This option should be disabled only if you register SIP Phones directly with DID Logic.

Tip. If you prefer to manage Caller IDs locally on FreeSWITCH, please be sure to either inform your account manager or email support@didlogic.com to have this feature enabled on your account (see Configuring Caller ID Passthrough for more details).

Click the Create button to complete the configuration.

Once SIP Account is created, you will see a 5-digit number that you should use as Username to authenticate your SIP device with DID Logic.

Setting up inbound routing

There are 3 options for the inbound call routing:

• SIP device: incoming calls will be routed to the respective SIP account if there is an active registration on any of the DID Logic gateways
• SIP URI (user_part@domain_part): incoming calls will be routed to the specified SIP URI address
• Phone: incoming calls will be forwarded to a PSTN number

Note:
- Multiple destinations can be configured for a single DID to ensure failover or provide call coverage options
- You can choose the preferred call distribution algorithms: ‘sequential’ or ‘ring all’

If registration is successful, you will see ‘REGED’ in the “State” column. Please note that if the SIP gateway is not registered, incoming calls will not be routed to your FreeSWITCH.

Configuring Inbound and Outbound calling for IP-authenticated SIP devices

You can enable IP Authentication for your SIP Account. In this case, DID Logic gateways will accept all outbound calls from the specified IP address without further authentication. To setup IP Authentication, please contact support at support@didlogic.com and provide a public IP Address you will be sending outbound calls from.

Note:
- SIP Accounts with IP Authentication enabled cannot be used as the destination assigned to DIDs on the portal. SIP URI addresses must be used instead.

Steps to configure incoming call routing to the SIP URI:

1. Log in to you DID Logic account
2. Go to the “PURCHASED” tab and click on “Edit”
3. Select "SIP URI" from the drop-down list in the “Forward to:“ field
4. Enter a valid SIP URI using the following format: +[did]@<Destination>:port;transport=type,
   where:
   • Destination is the fully qualified domain name or IP address of your FreeSWITCH
   • The port is the SIP connection port used by FreeSWITCH
   • The type could be either udp, tcp, or tls. The default transport is udp.
5. Click “Add” to set up the destination for the DID number

For this guide, let's use 61394607320@123.123.123.123:5060 as SIP URI (in this example, FreeSWITCH's IP Address is 123.123.123.123):

Let's adjust FreeSWITCH settings:

1. Update vars.xml /usr/local/freeswitch/conf/vars.xml:
add the line
<X-PRE-PROCESS cmd="set" data="outbound-sip-ip=123.123.123.123"/>
edit the lines
<XPREPROCESS cmd="stun-set" data="external_rtp_ip=123.123.123.123"/>
<XPREPROCESS cmd="stun-set" data="external_sip_ip=123.123.123.123"/>

2. Uncomment the lines for “ext-rtp-ip” and “ext-sip-ip” in /usr/local/freeswitch/conf/sip_profiles/external.xml and set them as a reference to the variables in vars.xml:
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
<param name="ext-sip-ip" value="$${external_sip_ip}"/>

3. Create an external SIP profile
/usr/local/freeswitch/conf/sip_profiles/external/didlogic_ip.xml

<include>
<gateway name="didlogic_ip">
<param name="proxy" value="sip.se.didlogic.net"/>
<param name="register" value="false"/>
<param name="context" value="didlogic_ip"/>
<param name="caller-id-in-from" value="true"/>
<param name="sip_cid_type" value="none"/>
<param name="retry-seconds" value="30"/>
<param name="expire-seconds" value="600"/>
</gateway>
</include>

4. Add the following line to the dialplan file:
<action application="set" data="sip_from_host=${outbound-sip-ip}"/>
and change the content as follows:

<?xml version="1.0" encoding="UTF-8"?>
<include>
    <context name="didlogic_ip">
        <extension name="unloop">
            <condition field="$${unroll_loops}" expression="^true$" />
            <condition field="${sip_looped_call}" expression="^true$">
                <action application="deflect" data="${destination_number}" />
            </condition>
        </extension>
        <extension name="out-didlogic_ip">
            <condition field="destination_number" expression="^(\d{10,15})$">
                <action application="set" data="sip_from_host=${outbound-sip-ip}"/>
                <action application="bridge" data="sofia/gateway/didlogic/$1"/>
                <action application="export" data="sip_from_uri=${sip_from_uri}"/>
                <action application="hangup"/>
            </condition>
        </extension>
        <extension name="in-didlogic">
            <condition field="destination_number" expression="^(61394607320)$">
                <action application="bridge" data="user/1000@$${domain}"/>
                <action application="hangup"/>
            </condition>
        </extension>
        <extension name="local">
            <condition field="destination_number" expression="^\d{4}$">
                <action application="answer" />
                <action application="sleep" data="1000" />
                <action application="bridge" data="user/${destination_number}@$${domain}" />
                <action application="hangup" />
            </condition>
        </extension>
    </context>
</include>

Now outbound calls should work.

Configuring Caller ID Passthrough

When you configure SIP Account settings on the DID Logic portal, you normally assign one of the purchased DID numbers as Caller ID.

If you want to use your own A-numbers and send them directly from FreeSWITCH, you need to contact your account manager or send an email to support@didlogic.com first and request Caller ID passthrough to be activated. Note that this feature may be unavailable depending on individual eligibility.

Once your request is approved and the feature is enabled, you need to remove the Caller ID assigned to your SIP Account to allow Custom Caller ID passthrough.

Updating SIP Account settings

Log in to the DID Logic Portal, go to the SIP section, and click Edit on the SIP Account. Remove Caller ID and click Update:

You should see your SIP Account updated:

Tip. If there is no Caller ID associated with the SIP Account while the Caller ID passthrough is disabled, your outbound calls will be passed over the public network with the Anonymous Caller ID. Anonymous calls tend to be routinely deprioritized and may even be barred by carriers.

Configuring Custom Caller ID on FreeSWITCH

1. Update your external SIP profile with new parameters:
   /usr/local/freeswitch/conf/sip_profiles/external/didlogic.xml
   

   <include>
   <gateway name="didlogic">
   <param name="username" value="39554"/>
   <param name="password" value="YOUR_SIP_ACCOUNT_PASSWORD"/>
   <param name="proxy" value="sip.se.didlogic.net"/>
   <param name="expire-seconds" value="600"/>
   <param name="register" value="true"/>
   <param name="retry-seconds" value="30"/>
   <param name="context" value="didlogic"/>
   <param name="caller-id-in-from" value="true"/>
   <param name="sip_cid_type" value="none"/>
   </gateway>
   </include

2. You can set Caller ID in the extension configuration file using the option ‘effective_caller_id_number’:
   /usr/local/freeswitch/conf/directory/default/1000.xml
   

   <include>
    <user id="1000">
      <params>
       <param name="password" value="YOUR_EXTENSION_PASSWORD"/>
       <param name="vm-password" value="1000"/>
      </params>
      <variables>
       <variable name="toll_allow" value="domestic,international,local"/>
       <variable name="accountcode" value="1000"/>
       <variable name="user_context" value="didlogic"/>
       <variable name="effective_caller_id_name" value="CALLER_ID_NAME"/>
       <variable name="effective_caller_id_number" value="61272534418"/>
       <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>
       <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
       <variable name="callgroup" value="techsupport"/>
      </variables>
    </user>
   </include>

Setting up failover routing

To ensure business continuity and safeguard against potential service interruptions, it's crucial to consider SIP Trunk failover scenarios.

This guide offers configuration options for failover mechanisms for both incoming and outgoing calls.

Inbound call failover

To protect against inbound call disruptions, we’ll use dual registrars. First, let’s allow FreeSWITCH to process incoming calls from both regional gateways.

Create failover profile didlogic_fo.xml in the directory containing external SIP profiles
usr/local/freeswitch/conf/sip_profiles/external/didlogic_fo.xml:

<include>
<gateway name="didlogic_fo">
<param name="username" value="39554"/>
<param name="password" value="<YOUR_SIP_ACCOUNT_PASSWORD>"/>
<param name="proxy" value="sip.nl.didlogic.net"/>
<param name="expire-seconds" value="600"/>
<param name="register" value="true"/>
<param name="retry-seconds" value="30"/>
<param name="context" value="didlogic"/>
<param name="caller-id-in-from" value="true"/>
<param name="sip_cid_type" value="none"/>
</gateway>
</include>

Outbound call failover

Update the dialplan configuration for outbound calls by adding the following lines:

<action application="set" data="continue_on_fail=true"/>
<action application="bridge" data="sofia/gateway/didlogic_fo/$1"/>

usr/local/freeswitch/conf/dialplan/didlogic.xml:

<?xml version="1.0" encoding="UTF-8"?>
  <include>
    <context name="didlogic">
    <extension name="unloop">
      <condition field="$${unroll_loops}" expression="^true$" />
      <condition field="${sip_looped_call}" expression="^true$">
        <action application="deflect" data="${destination_number}" />
      </condition>
    </extension>

   <extension name="in-didlogic">
      <condition field="destination_number" expression="^(61394607320)$">
        <action application="bridge" data="user/1000@$${domain}"/>
        <action application="hangup"/>
       </condition>
    </extension>

    <extension name="out-didlogic">
      <condition field="destination_number" expression="^(\d{10,15})$">
        <action application="bridge" data="sofia/gateway/didlogic/$1"/>
        <action application="set" data="continue_on_fail=true"/>
        <action application="bridge" data="sofia/gateway/didlogic_fo/$1"/>
        <action application="hangup"/>
      </condition>
   </extension>
  </context>
</include>

Common issues

Common issues with outbound calls