[Freeswitch-dev] call forward howto

[Anthony Minessale]

all X-headers are made into variables for you
route the call to the "info" app to see a dump of all the variables for that
call.

If there is some other standard header that we do not already supply that
you would like to see as a variable let us know and we can consider it.


On Sun, Apr 6, 2008 at 5:36 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> See brian's example on the list for something that might handle your
> situation.
>
> I'm not trying to explain away the need for what you want I just want to
> make sure you understand
> how we have things designed.
>
> You may want to join our irc channel for more realtime interaction.
> #freeswitch on irc.freenode.net
>
>
>
> On Sun, Apr 6, 2008 at 5:23 PM, kokoska rokoska <kokoska.rokoska at post.cz>
> wrote:
>
> >
> > 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
> >
> >
> > _______________________________________________
> > Freeswitch-dev mailing list
> > Freeswitch-dev at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > http://www.freeswitch.org
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20080406/5b20f853/attachment.html