[Freeswitch-users] SIP incoming call routing

[Saurabh Aggarwal]

Yes, but there is no DID in my system for incoming calls. I have users dynamically registering gateways, and calls coming in to SIP ids that they have used to register.
 
-Saurabh
 
 



Date: Wed, 29 Oct 2008 15:12:28 +0530From: talk2ram at gmail.comTo: freeswitch-users at lists.freeswitch.orgSubject: Re: [Freeswitch-users] SIP incoming call routing
On Wed, Oct 29, 2008 at 2:52 PM, Saurabh Aggarwal <saurabh_aggarwal at hotmail.com> wrote:

We are using freeswitch as a SIP proxy, where we are letting people register with freeswitch, and in-turn we do the SIP registration for them to "arbitrary" sip servers (as requested by users) - each user gets his own sip gateway in the freeswitch configuration. Then they can make outgoing calls and calls are routed through their specific SIP gateway. Now the problem is that when a call is received from one of these SIP registrations, it hits the public.xml where I can't seem to figure out how to get the SIP gateway information from which it came in. The SIP gateway name actually contains the information where it should be routed to. Any ideas on how to approach this problem? Question - is it possible to do it in the dialplan (dynamic) or do we have to write an application to do this mapping? -Saurabh
 
have you looked at this example
 
http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Receiving_an_inbound_call_from_a_Gateway
 
ram
_________________________________________________________________
When your life is on the go—take your life with you.
http://clk.atdmt.com/MRT/go/115298558/direct/01/
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20081029/afc05932/attachment-0002.html