[Freeswitch-users] SIP Trunk with Private Static IP

David Ponzone david.ponzone at gmail.com
Wed Jun 23 15:45:36 PDT 2010


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
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
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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 at 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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