Provider Configuration: SIPRoutes

From FreeSWITCH Wiki
Jump to: navigation, search

At time of this posting Siproutes provides outbound trunks only.


     <gateway name="SIPRoutes">
     <param name="username" value="user"/>
     <param name="password" value="pass"/>
     <param name="proxy" value="72.15.219.140"/>
     <param name="expire-seconds" value="800"/>
     <param name="register" value="false"/>
     <param name="retry-seconds" value="60"/>
     <param name="context" value="public"/>
     <param name="caller-id-in-from" value="true"/>
     <param name="sip_cid_type" value="pid"/>
     <param name="extension-in-contact" value="false"/>
     <param name="supress-cng" value="true"/> 
   </gateway>

SIPRoutes uses 11 digit dialing

   <extension name="Siproutes" >
  <condition field="destination_number" expression="^\+?1?(\d{10})$" >
             <action application="set" data="sip_h_X-Tag=" />
      <action application="set" data="call_direction=outbound" />
      <action application="set" data="hangup_after_bridge=true" />
      <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}" />
     <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}" />
      <action application="set" data="inherit_codec=true" />
      <action application="bridge" data="sofia/gateway/SIPRoutes/1$1" />
  </condition>
  </extension>

According to Avi Marcus on the mailing list, 04/24/14 They also recommend setting <param name="auto-rtp-bugs" value="clear"/> because of sonus see https://wiki.freeswitch.org/wiki/RTP_Issues#DTMF_Problems