[Freeswitch-users] SIP incoming call routing
anthony.minessale at gmail.com
Wed Oct 29 10:46:44 PDT 2008
whatever 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 <
saurabh_aggarwal at hotmail.com> 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.
> Date: Wed, 29 Oct 2008 15:12:28 +0530
> From: talk2ram at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: 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?
> have you looked at this example
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Anthony Minessale II
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