Thanks - that does work to an extent.
Now the problem is that not all gateways would allow "arbitrary" extensions.
E.g. AIM Callout - it *requires* that the extension/caller-id be your aim
username.
-Saurabh
Date: Wed, 29 Oct 2008 12:46:44 -0500From: [EMAIL PROTECTED]: [EMAIL
PROTECTED]: Re: [Freeswitch-users] SIP incoming call routingwhatever you put in
the "extension" param in the gateway should control what destination_number it
has in the inbound call. you can also do your regex in your dialplan on any of
the info in the sip packet besides destination number if you wish.
On Wed, Oct 29, 2008 at 4:52 AM, Saurabh Aggarwal <[EMAIL PROTECTED]> wrote:
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: [EMAIL PROTECTED]: [EMAIL
PROTECTED]: Re: [Freeswitch-users] SIP incoming call routing
On Wed, Oct 29, 2008 at 2:52 PM, Saurabh Aggarwal <[EMAIL PROTECTED]> 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
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