[Freeswitch-users] RE- Gateway failover question - SIP INVITE still be sent to the failed gateway after failover within the dialplan takes place

26 Aug 2015

      To FreeSWITCH Users,

I wondered if someone could assist me regarding failover when a gateway goes down.  FreeSWITCH version 1.4.20 (current production release).

The requirement from my SIP Trunk provider is to failover in about 4 to 5 seconds.

I have tried the following inside my dialplan:

      <action application="set" data="hangup_after_bridge=true"/>
      <action application="set" data="call_timeout=4"/>
      <action application="bridge" data="sofia/gateway/SIPprovider1/xxxxxxxxxx;user=phone|sofia/gateway/ SIPprovider2/xxxxxxxxxx;user=phone"/>
      <action application="sleep" data="1000"/>
      <action application="bridge" data="sofia/gateway/SIPprovider2/xxxxxxxxxx;user=phone"/>

Where xxxxxxxxxx is a valid phone number.

SIPprovider1 would be down (ie. SBC switched off) and SIPprovider2 would be working OK.  Obviously these are setup within the FreeSWITCH /conf/sip_profiles/external/myprovider.xml.

I have experimented with other various combinations:

      <action application="set" data="call_timeout=4"/>
      <action application="bridge" data="sofia/gateway/SIPprovider1/+44xxxxxxxxxx;user=phone|sofia/gateway/ SIPprovider2/+44xxxxxxxxxx;user=phone"/>
      <action application="sleep" data="1000"/>
      <action application="bridge" data="sofia/gateway/SIPprovider2/+44xxxxxxxxxx;user=phone"/>

or

      <action application="set" data="call_timeout=4"/>
      <action application="bridge" data="sofia/gateway/SIPprovider1/+44xxxxxxxxxx;user=phone|sofia/gateway/ SIPprovider2/+44xxxxxxxxxx;user=phone"/>

or

      <action application="set" data="hangup_after_bridge=true"/>
      <action application="set" data="call_timeout=4"/>
      <action application="bridge" data="sofia/gateway/SIPprovider1/+44xxxxxxxxxx;user=phone"/>
      <action application="sleep" data="1000"/>
      <action application="bridge" data="sofia/gateway/SIPprovider2/+44xxxxxxxxxx;user=phone"/>

The issue I am having is the 4 to 5 second switch to SIPprovider2 does work, however for some reason the SIP INVITE messages are still being sent to SIPprovider1 after the 4 seconds.  I need the SIP INVITE messages to STOP after 4 seconds to SIPprovider1 (which is down).

Any ideas would be most appreciated.

Kind Regards,

Andrew Keil
Visytel Pty Ltd