[Freeswitch-users] almost there: Easy question, I hope just need to proxy port 5060 to 5080

Michael Scheidell michael.scheidell at secnap.com
Thu Aug 26 03:09:42 PDT 2010


almost there, but the ../sip_profiles ../public ../default ../dialplan 
directories have me confused, especially since all these files 'look the 
same'. (almost)
I must REALLY be dense:

if I just want to bridge (since the provider won't ever send me anything 
but the did's they are getting paid for):

so: just TWO files?

cd /usr/local/freeswitch/conf

in ../dialplan/public/from_provider.xml I have: (do I need <include> 
..... </include> if this in in a file? tried both ways.)

<extension name="from_provider">
<condition field="network_addr" expression="68.100.226.97"/>
<condition field="destination_number" expression="^(.*)$">
<action application="bridge" data="sofia/gateway/sipxecs/$1"/>
</condition>
</extension>

and in ../sip_profiles/external/sipxecs.xml I have: (I don't need 
authentication/registration).. im local :-)

<include>
<gateway name="sipxecs">
<param name="username" value="username"/>
<param name="password" value="password"/>
<param name="from-user" value="fromuser"/>
<param name="from-domain" value="192.168.0.2"/>
<param name="proxy" value="192.168.0.2"/>
<param name="expire-seconds" value="600"/>
<param name="register" value="false"/>
<param name="retry-seconds" value="30"/>
<param name="context" value="public"/>
</gateway>
</include>

I must really be an idiot, because now, I don't even get anything in the 
logs.  tcpdump shows INVITES from the provider.
(I added <include /> to the from_provider.xml just to see)

../log/freeswitch.log doesn't show anything after the restart stuff.  
and when I had it wrong, at least I logged the trys and 480 'try later' 
error.

taking a break for breakfast.. its 6am. back in 2 hours.

I want a sample setup.
I dial the 301 number, come in freeswitch, is gatewayed to my sipxecs.  
I get audio!  I get DTMF, I can internally transfer,
(oh, and when I finally get this to work. how do I send this to port 
5080 on sipx? so I can transfer from AA and/ or CFNA?)  I don't need the 
outbound gateway (yet)

(to complicate matters, this is the freeswitch package on pfsense.  no, 
not the 'new one', but the one I can install via the pfsense gui.  and 
restarts are slow)


On 8/26/10 1:49 AM, jesse wrote:
> I feel Michael's idea should also work:  If call is from X.X.X.X, then
> bridge the call with the same
> destination_number ($1) to sipxecs gateway.
>
> <extension name="from_provider">
>      <condition field="network_addr" expression="X.X.X.X"/>
>      <condition field="destination_number" expression="^(.*)$">
>        <action application="bridge" data="sofia/gateway/sipxecs/$1"/>
>      </condition>
>    </extension>
>
>
> On Wed, Aug 25, 2010 at 4:33 PM, David Ponzone<david.ponzone at ipeva.fr>  wrote:
>    
>> It is right.
>> You just transfer to another dialplan (context) in order to make it more
>> readable.
>> You'll be able to add all your rules in the provider context, because only
>> calls coming from provider X.X.X.X will go into this dialplan.
>> Remember: extension name has absolutely NO importance in FS. It's just a
>> label.
>> So to sum it all:
>> if call from X.X.X.X, transfer to context provider, with the same
>> destination_number ($1).
>> In context provider, if destination_number is XXX, then bridge the call to
>> sipxecs.
>> 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
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>      

-- 
Michael Scheidell, CTO
o: 561-999-5000
d: 561-948-2259
ISN: 1259*1300
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