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FreeSwitch

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FreeSwitch Configuration

Termination

You may need to add from domain param set to voip.ms for termination to work.

<param name="from-domain" value="voip.ms"/>

Config

 <include>
  <gateway name="voipms">
    <!-- Replace the values below with your Voip.ms username and password. -->
    <param name="username" value="your_username" />
    <param name="password" value="your_password" />
    <!-- This gateway could be different depending on which switch you are on -->
    <param name="proxy" value="montreal|houston|newyork|etc.voip.ms" />
    <param name="realm" value="voip.ms" />
    <!-- This should be set to "true" for registration based -->
    <param name="register" value="true" />
    <!-- Voip.ms requires the Remote-Party-Identity Header to be set in the Sip invite for Caller-ID to work right
        DON'T FORGET TO REMOVE ANY CALLER ID INFO IN http://voip.ms->Main Menu->Account Settings->General->CallerID Number
    -->
    <param name="sip_cid_type" value="rpid" /> 
    <!--Setting in one place is much easier than everywhere you may bridge. You can do this since 2010 Sept 27 
       http://jira.freeswitch.org/browse/FS-2722
    -->
  </gateway>
 </include>
 

Caller ID

Voip.ms requires the Remote-Party-Identity Header to be set in the Sip invite. Use:

<param name="sip_cid_type" value="rpid" /> 

Errors

  1. destination_number (inbound public) is getting set as my Sip_profile username.
    • if you have a subaccount set up, make sure "Device Type" is set to "Asterisk, ip pbx, gateway or voip switch" and not "ata device, ip phone or soft phone". Once this is fixed, destination_number will be the correct DID number.

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