[Freeswitch-dev] call forward howto

[kokoska rokoska]

Thank you, Anthony, for your answer. I will write my comments in the 
e-mail body.

Anthony Minessale napsal(a):
> The transfer app sends the same call to some other extension at some 
> other dialplan within the switch.
> We are not a sip specific switch so we have to abstract many concepts.
> 

Yes, I know it. But I just ask :-)
BTW: Very similar messaging concept is IMO used not only in SIP but in 
ISUP (and thus in real telephony world) too.


> If you are trying to redirect the call we have 2 ways
> the redirect application
> 
> <action application="redirect" data="sip:user at host"/>
> This will result in a 302 redirect in the case of a sip channel
> 

I saw this app. But I want to forward the call, not redirect.


> <action application="deflect" data="sip:user at host"/>
> This will result in a REFER to the specified uri
> 

REFER is IMO used for replacing of one leg of call with another one 
(RFC3515), i.e. for "transfer".


> That is the only 2 ways we support, if you are looking to divert like a 
> proxy does

Yes, this is what I have to do, because I need to be "transparent" for 
calls come from PSTN and than go back to PSTN too (ISUP->SIP->ISUP).

> remember we are only a b2bua and do not perform proxy features,

OK. This is big advantage for me, because I need B2BUA :-)
But what I want is possible with B2BUA. Even with Asterisk, but only 
with very ugly hacks I hope I am avoid with Freeswitch.

> see 
> openser for that.
> 

I'm intensive user of OpenSER. In my current setup it is used as 
loadbalancer, registrar, presence server etc. But I realy need B2BUA and 
ability to manipulate with media (with much more sophisticated way than 
SEMS does).


Thanks again for your replay, regards

kokoska.rokoska