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Buy a phone number from Wavix

Before placing or receiving a call, you need to have an active phone number on your Wavix account. If you already have one, you can skip this section.
  1. Sign in to your Wavix account.
  2. Click on Buy under Numbers & trunks in the top menu.
  3. Select the country and region where you wish to purchase a number.
  4. Choose one or more numbers and click Buy now.
  5. Review the Cart and complete checkout.
Some numbers require proof of local address or other documents before activation. Upload any requested documents and wait for the Wavix Provisioning team to approve them before the numbers become active.

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Create a SIP Trunk on the Wavix platform

  1. In the top menu, go to Numbers & Trunks → Trunks.
  2. Click Create new.
  3. Under Authentication Method, select Digest.
  4. Enter a SIP trunk name, set a SIP trunk password, and choose one of your numbers as the Caller ID.
  5. (Optional) Configure additional limits. If not specified, global account limits apply:
    • Max outbound call duration
    • Max number of simultaneous calls via the SIP trunk
    • Max call cost
  6. Click Create.
Once the SIP trunk is successfully created, it will appear in your account’s SIP trunks list. Configure SIP trunk screenshot
Your 5-digit SIP trunk username is generated automatically and shown in the SIP trunk ID column.

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Basic SIP trunk setup on FreeSWITCH

For this guide, we use the following Wavix gateways:
  • Domain name: us.wavix.net; IP address: 209.58.144.243 as the primary gateway.
  • Domain name: nl.wavix.net; IP address: 95.211.82.14 as a secondary gateway.
Choose the primary and backup gateways that offer the lowest ping from your premises. The full list of Wavix regional gateways can be found at the bottom of the page: https://app.wavix.com/trunks.

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Set up outbound calls

To configure inbound and outbound calls on your FreeSWITCH server:
  1. Navigate to /usr/local/FreeSwitch/conf/sip_profiles/external
  2. Create a new file wavix.xml
<include>
    <gateway name="wavix">
        <param name="username" value="YOUR_SIP_TRUNK_ID"/>
        <param name="password" value="YOUR_SIP_TRUNK_PASSWORD"/>
        <param name="realm" value="us.wavix.net"/>
        <param name="register" value="true"/>
        <param name="context" value="wavix"/>
    </gateway>
</include>
  • The authentication realm must match the domain name of the regional Wavix gateway you would routing your calls over
  • Enter the SIP trunk ID and password you configured for the SIP trunk on the Wavix platform in the “username” and “password” variables respectively
After you create the file, assign the FreeSWITCH user as the owner so the application can access it: 
root@FreeSwitch:/usr/local/FreeSwitch/conf/sip_profiles/external/# chown FreeSwitch:FreeSwitch wavix.xml
  1. Navigate to /usr/local/FreeSwitch/conf/dialplan/
  2. Create a new dialplan file wavix_dialplan.xml
<?xml version="1.0" encoding="UTF-8"?>
<include>
    <context name="wavix">
        <extension name="unloop">
            <condition field="$${unroll_loops}" expression="^true$" />
            <condition field="${sip_looped_call}" expression="^true$">
                <action application="deflect" data="${destination_number}" />
            </condition>
        </extension>
        <extension name="in-wavix">
            <condition field="destination_number" expression="^(16468097065)$">
                <action application="bridge" data="user/1000@$${domain}"/>
                <action application="hangup"/>
            </condition>
        </extension>
        <extension name="out-wavix">
            <condition field="destination_number" expression="^(\d{10,15})$">
                <action application="bridge" data="sofia/gateway/wavix/$1"/>
                <action application="export" data="sip_from_uri=${sip_from_uri}"/>
                <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}@$${domain}" />
                <action application="hangup" />
            </condition>
        </extension>
    </context>
</include>
expression=- a regular expression matching dialed number. In this example, the rule will work if the dialed phone number contains 10 to 15 digits.
After you create the file, give FreeSWITCH access so it can read it. All destination phone numbers must be in E.164 international format. E.164 numbers can have up to fifteen digits and are usually formatted as [+][country code][subscriber number including area code]. For example, a US number in E.164 format is +16561223344. Calls to numbers without a country code or with national access codes will be rejected by Wavix. Below are typical examples of incorrectly formatted phone numbers:
CountryNumberReasonNumber in E.164 format
US6561223344No country code+16561223344
UK020 1122 3344No country code, national access code with 0+442011223344
Switzerland0041797000777Leading 00 international prefix+41797000777
  1. Navigate to /usr/local/FreeSwitch/conf/directory/default/
  2. Modify extension configuration file 1000.xml
<include>
    <user id="1000">
        <params>
            <param name="password" value="SET_EXTENSION_PASSWORD"/>
            <param name="vm-password" value="1000"/>
        </params>
        <variables>
            <variable name="toll_allow" value="domestic,international,local"/>
            <variable name="accountcode" value="1000"/>
            <variable name="user_context" value="wavix"/>
            <variable name="effective_caller_id_name" value="CALLER_ID_NAME"/>
            <variable name="effective_caller_id_number" value="16468097065"/>
            <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>
            <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
            <variable name="callgroup" value="support"/>
        </variables>
    </user>
</include>
The value of the user_context variable must match the name of the gateway configuration you created in the external SIP profile directory (without the .xml extension).
  1. Navigate to /usr/local/FreeSwitch/conf/autoload_configs/default/
  2. Define RTP port range to 10000-20000 in switch.conf.xml:
<param name="rtp-start-port" value="10000"/>
<param name="rtp-end-port" value="20000"/>
  1. To reload the configuration files, run:
fs_cli -x reloadxml

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Enable IP authentication

Wavix will accept all calls originating from a particular IP address, without requiring login info, when IP authentication is enabled. You must have a dedicated IP for this option to work correctly. Wavix SIP trunk ID authentication Follow these steps to enable IP authentication on the Wavix platform:
  1. Select Trunks under Numbers & trunks in the top menu
  2. Click the More icon for the desired SIP trunk and click Edit
  3. Select IP Authentication under the Authentication method
  4. Put public IP address of your endpoint
  5. Click Submit
  6. Click Save to apply changes
Wavix support team will review your request and enable IP Authentication on the SIP trunk.
  • When IP authentication is enabled on a SIP trunk, it cannot be used as a destination for a DID. You must use a SIP URI instead.
  • After submitting the IP authentication request, you will not be able to update your IP address or change the authentication method. By default, an IP address can only be mapped to a single SIP trunk. If you need to have several Wavix SIP trunks sharing the same IP address, contact [email protected].
Once your request is approved, IP authentication will be activated on your Wavix SIP trunk, and you can configure it on FreeSWITCH:
  1. Navigate to /usr/local/FreeSwitch/conf/
  2. Update the vars.xml file with the lines below:
<X-PRE-PROCESS cmd="set" data="outbound-sip-ip=123.123.123.123"/>
<XPREPROCESS cmd="stun-set" data="external_rtp_ip=123.123.123.123"/>
<XPREPROCESS cmd="stun-set" data="external_sip_ip=123.123.123.123"/>
  1. Uncomment the lines for ext-rtp-ip and ext-sip-ip in the global configuration file external.xml and set them to reference the variables from vars.xml:
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
<param name="ext-sip-ip" value="$${external_sip_ip}"/>
  1. Navigate to /usr/local/FreeSwitch/conf/sip_profiles/external/
  2. Create a new SIP profile wavix_ip.xml
<include>
    <gateway name="wavix_ip">
        <param name="proxy" value="us.wavix.net"/>
        <param name="register" value="false"/>
        <param name="context" value="wavix_gateway_ip"/>
        <param name="retry-seconds" value="30"/>
        <param name="expire-seconds" value="600"/>
    </gateway>
</include>
  1. Add the following line to the dialplan:
<action application="set" data="sip_from_host=${outbound-sip-ip}"/>

<?xml version="1.0" encoding="UTF-8"?>
<include>
    <context name="wavix_ip">
        <extension name="unloop">
            <condition field="$${unroll_loops}" expression="^true$" />
            <condition field="${sip_looped_call}" expression="^true$">
                <action application="deflect" data="${destination_number}" />
            </condition>
        </extension>
        <extension name="out-wavix_ip">
            <condition field="destination_number" expression="^(\d{10,15})$">
                <action application="set" data="sip_from_host=${outbound-sip-ip}"/>
                <action application="bridge" data="sofia/gateway/wavix/$1"/>
                <action application="export" data="sip_from_uri=${sip_from_uri}"/>
                <action application="hangup"/>
            </condition>
        </extension>
        <extension name="in-wavix">
            <condition field="destination_number" expression="^(16468097065)$">
                <action application="bridge" data="user/1000@$${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>
  1. Once you’ve created the file, assign the FreeSWITCH user as the owner:
root@FreeSwitch:/usr/local/FreeSwitch/conf/sip_profiles/external/# chown FreeSwitch:FreeSwitch wavix_ip.xml
Configuration is complete and outbound calls should work now.

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Placing outbound calls with caller ID passthrough

Typically, you use one of the DID numbers you purchased as the caller ID for the SIP trunk. Wavix SIP trunk Caller ID Passthrough Alternatively, you can enable the caller ID passthrough option on the Wavix platform. This allows you to send your own A-numbers directly from FreeSWITCH:
  1. Select Trunks under Numbers & trunks in the top menu
  2. Click on the three dots on the right hand side and select the Edit option
  3. Select Passthrough under the Caller ID
  4. Click Activate
  5. Click Save to apply changes
Wavix support team will review your request and enable Caller ID Passthrough on the SIP trunk.

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Configure custom caller ID on FreeSWITCH

Once your request is approved, the caller ID passthrough option will be activated on your Wavix SIP trunk and you can send your own caller IDs from FreeSWITCH:
  1. Navigate to /usr/local/FreeSwitch/conf/sip_profiles/external/
  2. Update the outbound SIP gateway configuration file wavix.xml:
<include>
    <gateway name="wavix">
        <param name="username" value="SIP_TRUNK_ID"/>
        <param name="password" value="SIP_TRUNK_PASSWORD"/>
        <param name="realm" value="us.wavix.net"/>
        <param name="register" value="true"/>
        <param name="context" value="wavix"/>
        <param name="caller-id-in-from" value="true"/>
        <param name="sip_cid_type" value="none"/>
    </gateway>
</include>
  1. Assign the caller ID to the variable effective_caller_id_number in the extension configuration file:
<include>
    <user id="1000">
        <params>
            <param name="password" value="SET_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="wavix"/>
            <variable name="effective_caller_id_name" value="CALLER_ID_NAME"/>
            <variable name="effective_caller_id_number" value="16468097065"/>
            <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>
            <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
            <variable name="callgroup" value="support"/>
        </variables>
    </user>
</include>

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Set up inbound calls

Registering your SIP trunk with one of the regional Wavix gateways is necessary (for digest authentication only) to receive inbound calls. Register and send SIP traffic to regional gateways for low latency access. Read our FAQ.
Amsterdam, NLDallas, USASingapore, SGAustralia, AU
nl.wavix.netus.wavix.netsg.wavix.netau.wavix.net
Select a regional gateway with the lowest latency to your device (in this example, us.wavix.net):
  1. Navigate to /usr/local/FreeSWITCH/conf/sip_profiles/external/
  2. Create or update SIP external profile wavix.xml:
<include>
    <gateway name="wavix">
        <param name="username" value="YOUR_SIP_TRUNK_ID"/>
        <param name="password" value="YOUR_SIP_TRUNK_PASSWORD"/>
        <param name="realm" value="us.wavix.net"/>
        <param name="register" value="true"/>
        <param name="context" value="wavix"/>
        <param name="caller-id-in-from" value="true"/>
        <param name="sip_cid_type" value="none"/>
    </gateway>
</include>
  • Each regional gateway has a unique authentication realm that corresponds to its domain name.
  • Enter the SIP trunk ID and password you configured for the SIP trunk on the Wavix platform in the username and password parameters.
You can check the status of your SIP trunk registration using the "fs_cli -x "sofia status gateway wavix" | grep State" command: 
root@freeswitch:~# fs_cli -x "sofia status gateway wavix" | grep State
PringState   0/0/0
State        REGED
If registration is successful, you will see REGED in the “State” column. If the SIP gateway is not registered, incoming calls will not be routed to your FreeSWITCH. Wavix DID configuration To configure the destination for inbound calls on the Wavix platform:
  1. Select My numbers under Numbers & trunk in the top menu
  2. Click on the three dots on the right-hand side and select the Edit number option.
  3. Select the destination trunk in the Destination section
  4. Click Add to add the destination for the DID
  5. Click Save to apply changes
Configuration is complete and inbound calls should work now.

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SIP trunk redundancy

Considering SIP trunk redundancy is important to ensure business continuity and protect against service interruptions. This guide provides configuration options to ensure redundancy for both incoming and outgoing calls.

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Redundancy options for inbound calls

In case of a regional gateway failure, Wavix may originate inbound calls from any other gateway. You can find the full list of gateways at https://app.wavix.com/trunks. There are two options to prevent disruptions of inbound calls: set up a SIP URI destination for the DID number or enable dual SIP trunk registrations from FreeSWITCH. In this guide, we’ll use a SIP URI. Wavix DID configuration To configure a SIP URI destination on the Wavix platform:
  1. Select My numbers under Numbers & trunk in the top menu
  2. Click the three dots icon for the desired DID and click Edit number
  3. Select SIP URI under the Destination and enter URI in the format: DID@<Public IP of your endpoint>
  4. Click Add
  5. Click Save to apply changes
  6. Navigate to /usr/local/FreeSwitch/conf/dialplan/
  7. Specify your DID numbers in dialplan in <extension name="in-wavix"> extension
<extension name="in-wavix">
    <condition field="destination_number" expression="^(16468097065)$">
        <action application="bridge" data="user/1000@$${domain}"/>
        <action application="hangup"/>
    </condition>
</extension>

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Redundancy options for outbound calls

  1. Navigate to /usr/local/FreeSwitch/conf/sip_profiles/external/
  2. Create a new file wavix_fo.xml
<include>
    <gateway name="wavix_fo">
        <param name="username" value="YOUR_SIP_TRUNK_ID"/>
        <param name="password" value="YOUR_SIP_TRUNK_PASSWORD"/>
        <param name="proxy" value="nl.wavix.net"/>
        <param name="expire-seconds" value="600"/>
        <param name="register" value="true"/>
        <param name="retry-seconds" value="30"/>
        <param name="context" value="wavix"/>
        <param name="caller-id-in-from" value="true"/>
        <param name="sip_cid_type" value="none"/>
    </gateway>
</include>
  1. Update the dialplan for outbound calls by adding the following lines:
<action application="set" data="continue_on_fail=true"/>
<action application="bridge" data="sofia/gateway/wavix_fo/$1"/>
See a dialplan example below: 
<?xml version="1.0" encoding="UTF-8"?>
<include>
    <context name="wavix">
        <extension name="unloop">
            <condition field="$${unroll_loops}" expression="^true$" />
            <condition field="${sip_looped_call}" expression="^true$">
                <action application="deflect" data="${destination_number}" />
            </condition>
        </extension>
        <extension name="in-wavix">
            <condition field="destination_number" expression="^(16468097065)$">
                <action application="bridge" data="user/1000@$${domain}"/>
                <action application="hangup"/>
            </condition>
        </extension>
        <extension name="out-wavix">
            <condition field="destination_number" expression="^(\d{10,15})$">
                <action application="bridge" data="sofia/gateway/wavix/$1"/>
                <action application="export" data="sip_from_uri=${sip_from_uri}"/>
                <action application="set" data="continue_on_fail=true"/>
                <action application="bridge" data="sofia/gateway/wavix_fo/$1"/>
                <action application="hangup"/>
            </condition>
        </extension>
        <extension name="local">
            <condition field="destination_number" expression="^\d{4}$">
                <action application="answer" />
                <action application="sleep" data="1000" />
                <action application="bridge" data="user/${destination_number}@$${domain}" />
                <action application="hangup" />
            </condition>
        </extension>
    </context>
</include>

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Troubleshooting

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Inbound call issues

  • Make sure you have granted FreeSWITCH access to the XML files you have created.
  • Make sure that the context name in the dialplan, internal extension, and SIP profile extension are matching.

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Outbound calls issues

  • A 603 Declined response might be received when calling a destination with a per-minute rate higher than the max call rate set for your account. You can find your effective max call rate on the Trunks page. Contact [email protected] to request a change to your max call rate.
  • Another possible reason for 603 Declined responses is lack of active registration. You can check the registration status of the SIP trunk as outlined above.
  • Wrong destination number format. Make sure you dial the full international number (E.164): 19085594899 (US), 4408001218915 (UK). Dialing 9085594899 won’t work. Strip prefixes like 0, 00, or 011 in front of the dialed number.

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FreeSWITCH diagnostics

  • Verify SIP trunk registration with sofia status gateway <gateway-name> in the fs_cli command prompt. If registration is successful, you will see REGED in the “State” column. If the SIP gateway is not registered, incoming calls will not be routed to your FreeSWITCH.
  • To analyze SIP traffic, use the sngrep tool with appropriate filters