[Freeswitch-users] Having trouble establishing a call

[Mark Sobkow]

Andrew Thompson wrote:
> On Tue, Feb 09, 2010 at 12:46:54PM -0600, Mark Sobkow wrote:
>   
>> We're using Erlang to serve up the configurations to Freeswitch.  I've 
>> got things configured such that I can place a call from a SIP phone 
>> registered to extension 5000 to our "external" SIP provider (our 
>> Asterisk installation), but I can't place a call to extension 5001 from 
>> 5000.  Below is the trace log Freeswitch produces when I attempt to do so.
>>
>> Any suggestions as to what I should be looking at?  The directory seems 
>> to be getting served up correctly, as it provides the passwords both SIP 
>> softphones are using to register with Freeswitch.  I'd have thought that 
>> once they've registered with FS, the extension would automatically be 
>> recognized when an incoming call is placed or bridged, but such does not 
>> seem to be the case.
>>
>>     
>
> Looks like the other sip box (sofia/external/5001 at testsrv.marketel) is
> returning the failure code. Check the config on the other side? Don't
> you have a gateway setup for this other machine so you couls do
> sofia/gateway/testsrv.marketel/5001 instead?
>
> Andrew
>
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>   

Actually rats.marketel is our Asterisk box, which acts as our SIP 
trunk.  testsrv.marketel is the one running Freeswitch.

Am I maybe using an incorrect syntax in the dialplan for specifying that 
the call should be routed to a local extension on the Freeswitch box?  I 
thought the sofia/<profile>/<number>@<server> syntax is supposed to be 
used for placing any SIP calls, not just "remote" ones.