Baskar,

there are many inconsistencies/errors in the config you sent.

1) You can't have 2 default routes on a Unix box (well, you can, but the results will be unpredictable). The usual configuration is to have no default route for the LAN (eth0), or possibly some specific static routes, and a default route to the outside (eth1)
2) 10.10.10.3 is not part of the subnet on your eth0 (10.10.10.40/29), so you can't use that as a default route
3) 10.10.10.47 is the broadcast address of the subnet 10.10.10.40/29, so you can't use that. Perhaps 47 was a typo, that would also explain the issue above.
4) if 10.0.0.2 is the IP of your provider, you should really check that you can reach that with the routes you have (see 1 above). If you are running pure private on this box, I would recommend not setting any default route, but rather setting all the required static routes

David Ponzone  Direction Technique
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26

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Le 24/06/2010 à 00:22, Baskar a écrit :

Hi All,

I have set a freeswitch which has two ethernet cards, one is connected to local lan and the other is connected with our SIP providers trunk.Below is the config of both the ethernet cards

eth1 – SIP Provider IP: 172.16.61.27 MASK: 255.255.255.248 GW: 172.16.61.25
eht0 – Internal LAN IP: 10.10.10.47 MASK: 255.255.255.248 GW: 10.10.10.3

Dial plan for this SIP provider

  <extension name="sify">
    <condition field="destination_number" expression="^(\d{17}|\d{18|\d{19}})">
      <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>
      <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/>
      <action application="bridge" data="sofia/external/0109#001$1@10.0.0.2"/>
    </condition>
  </extension>

SIP Trunk settings from provider:

Protocol:   SIP
Prefix: 0109#001
Signaling IP: 10.0.0.2
Authentication:  IP based
Sip port :5060

There is no user account and password settings for this provider

I am able to dial out to local extension using this setting but am not able to make external calls through SIP trunk. The same settings are working fine with trixbox but it does not work with freeswitch.

We have another SIP account from one more SIP provider with a username and password and its working fine with freeswitch.Below is the DIal plan of another provider which works fine.
 

Dial Plan from the other provider

   <extension name="Pulse">
      <condition field="destination_number" expression="^(\d{11}|\d{12|\d{13}})">
      <action application="set" data="effective_caller_id_number=12223334444"/>
      <param name="caller-id-in-form" value="true"/>
      <action application="record_session" data="$${base_dir}/${destination_number}-${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
      <action application="bridge" data="sofia/gateway/7257029/$1"/>
      </condition>
   </extension>

Can you please help me to troubleshoot this issue.  Any pointers would be most appreciated.

--
Thanks with Regards,

N.Baskar

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