[Freeswitch-users] Loopback SIP invite to originator

[Jurijs Ivolga]

Hi,

I plan to use Freeeswitch as Media server for Kamailio and in my case
Freeswitch should receive call and loopback it back to Kamailio.

In my case I needed to make do not change: to, from and request-uri
headers. I achieved it by:

<action application="bridge"
data="[sip_from_uri=${sip_from_uri},sip_invite_to_uri=sip:${sip_to_uri}]sofia/internal/${sip_req_uri};transport=tcp;fs_path=sip:kamailioserver"/>

Thank you!

With kind regards,

Jurijs

2015-08-14 11:39 GMT+03:00 Stanislav Sinyagin <ssinyagin at gmail.com>:

> there's Kamailio and OpenSIPS for that, why do you need that on FreeSWITCH?
>
> On Wed, Aug 12, 2015 at 11:06 AM, Jurijs Ivolga <jurij.ivo at gmail.com>
> wrote:
> > Hi,
> >
> > I need to receive call on Freeswitch and Loopback it to original server
> from
> > where it came into, without changing any SIP header(to, from, SIP request
> > and etc). Is there easy way to do this?
> >
> > Thank you!
> >
> > With kind regards,
> >
> > Jurijs
> >
> > _________________________________________________________________________
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> http://www.freeswitchsolutions.com
>
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> http://confluence.freeswitch.org
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>
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