Loopback SIP invite to originator
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
If it is really a requirement to not change ANY headers, then no. On Wednesday, August 12, 2015, Jurijs Ivolga <jurij.ivo@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
What are you actually trying to accomplish? As stated, this doesn’t make much sense. -Michael From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Jurijs Ivolga Sent: Wednesday, August 12, 2015 3:06 AM To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Subject: [Freeswitch-users] Loopback SIP invite to originator 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
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@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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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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@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@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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (4)
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Jurijs Ivolga
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Michael Giagnocavo
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Michael Jerris
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Stanislav Sinyagin